Repo created

TMessagesProj/jni/voip/webrtc/api/OWNERS

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+hta@webrtc.org
+magjed@webrtc.org
+perkj@webrtc.org
+tommi@webrtc.org
+
+# For approvals that absolutely must be done on US Pacific time
+deadbeef@webrtc.org
+tkchin@webrtc.org
+
+per-file peer_connection*=hbos@webrtc.org
+
+per-file DEPS=mbonadei@webrtc.org
+
+per-file uma_metrics.h=kron@webrtc.org

TMessagesProj/jni/voip/webrtc/api/README.md

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+<!-- go/cmark -->
+<!--* freshness: {owner: 'hta' reviewed: '2021-01-01'} *-->
+
+# How to write code in the `api/` directory
+
+Mostly, just follow the regular [style guide](/g3doc/style-guide.md), but:
+
+* Note that `api/` code is not exempt from the “`.h` and `.cc` files come in
+  pairs” rule, so if you declare something in `api/path/to/foo.h`, it should be
+  defined in `api/path/to/foo.cc`.
+* Headers in `api/` should, if possible, not `#include` headers outside `api/`.
+  It’s not always possible to avoid this, but be aware that it adds to a small
+  mountain of technical debt that we’re trying to shrink.
+* `.cc` files in `api/`, on the other hand, are free to `#include` headers
+  outside `api/`.
+* Avoid structs in api, prefer classes.
+
+The preferred way for `api/` code to access non-`api/` code is to call
+it from a `.cc` file, so that users of our API headers won’t transitively
+`#include` non-public headers.
+
+For headers in `api/` that need to refer to non-public types, forward
+declarations are often a lesser evil than including non-public header files. The
+usual [rules](/g3doc/style-guide.md#forward-declarations) still apply, though.
+
+`.cc` files in `api/` should preferably be kept reasonably small. If a
+substantial implementation is needed, consider putting it with our non-public
+code, and just call it from the `api/` `.cc` file.
+
+Avoid defining api with structs as it makes harder for the api to evolve.
+Your struct may gain invariant, or change how it represents data.
+Evolving struct from the api is particular challenging as it is designed to be
+used in other code bases and thus needs to be updated independetly from its usage.
+Class with accessors and setters makes such migration safer.
+See [Google C++ style guide](https://google.github.io/styleguide/cppguide.html#Structs_vs._Classes) for more.
+
+If you need to evolve existent struct in api, prefer first to convert it into a class.

TMessagesProj/jni/voip/webrtc/api/adaptation/BUILD.gn

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+# Copyright(c) 2020 The WebRTC project authors.All Rights Reserved.
+#
+# Use of this source code is governed by a BSD - style license
+# that can be found in the LICENSE file in the root of the source
+# tree.An additional intellectual property rights grant can be found
+# in the file PATENTS.All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_source_set("resource_adaptation_api") {
+  visibility = [ "*" ]
+  sources = [
+    "resource.cc",
+    "resource.h",
+  ]
+  deps = [
+    "../../api:scoped_refptr",
+    "../../rtc_base:checks",
+    "../../rtc_base:refcount",
+    "../../rtc_base/system:rtc_export",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/adaptation/DEPS

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+specific_include_rules = {
+  "resource\.h": [
+    # ref_count.h is a public_deps of rtc_base:refcount. Necessary because of
+    # rtc::RefCountInterface.
+    "+rtc_base/ref_count.h",
+  ],
+}

TMessagesProj/jni/voip/webrtc/api/adaptation/resource.cc

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+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/adaptation/resource.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+const char* ResourceUsageStateToString(ResourceUsageState usage_state) {
+  switch (usage_state) {
+    case ResourceUsageState::kOveruse:
+      return "kOveruse";
+    case ResourceUsageState::kUnderuse:
+      return "kUnderuse";
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+ResourceListener::~ResourceListener() {}
+
+Resource::Resource() {}
+
+Resource::~Resource() {}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/adaptation/resource.h

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+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_ADAPTATION_RESOURCE_H_
+#define API_ADAPTATION_RESOURCE_H_
+
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class Resource;
+
+enum class ResourceUsageState {
+  // Action is needed to minimze the load on this resource.
+  kOveruse,
+  // Increasing the load on this resource is desired, if possible.
+  kUnderuse,
+};
+
+RTC_EXPORT const char* ResourceUsageStateToString(
+    ResourceUsageState usage_state);
+
+class RTC_EXPORT ResourceListener {
+ public:
+  virtual ~ResourceListener();
+
+  virtual void OnResourceUsageStateMeasured(
+      rtc::scoped_refptr<Resource> resource,
+      ResourceUsageState usage_state) = 0;
+};
+
+// A Resource monitors an implementation-specific resource. It may report
+// kOveruse or kUnderuse when resource usage is high or low enough that we
+// should perform some sort of mitigation to fulfil the resource's constraints.
+//
+// The methods on this interface are invoked on the adaptation task queue.
+// Resource usage measurements may be performed on an any task queue.
+//
+// The Resource is reference counted to prevent use-after-free when posting
+// between task queues. As such, the implementation MUST NOT make any
+// assumptions about which task queue Resource is destructed on.
+class RTC_EXPORT Resource : public rtc::RefCountInterface {
+ public:
+  Resource();
+  // Destruction may happen on any task queue.
+  ~Resource() override;
+
+  virtual std::string Name() const = 0;
+  // The `listener` may be informed of resource usage measurements on any task
+  // queue, but not after this method is invoked with the null argument.
+  virtual void SetResourceListener(ResourceListener* listener) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_ADAPTATION_RESOURCE_H_

TMessagesProj/jni/voip/webrtc/api/array_view.h

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+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_ARRAY_VIEW_H_
+#define API_ARRAY_VIEW_H_
+
+#include <algorithm>
+#include <array>
+#include <iterator>
+#include <type_traits>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/type_traits.h"
+
+namespace rtc {
+
+// tl;dr: rtc::ArrayView is the same thing as gsl::span from the Guideline
+//        Support Library.
+//
+// Many functions read from or write to arrays. The obvious way to do this is
+// to use two arguments, a pointer to the first element and an element count:
+//
+//   bool Contains17(const int* arr, size_t size) {
+//     for (size_t i = 0; i < size; ++i) {
+//       if (arr[i] == 17)
+//         return true;
+//     }
+//     return false;
+//   }
+//
+// This is flexible, since it doesn't matter how the array is stored (C array,
+// std::vector, rtc::Buffer, ...), but it's error-prone because the caller has
+// to correctly specify the array length:
+//
+//   Contains17(arr, arraysize(arr));     // C array
+//   Contains17(arr.data(), arr.size());  // std::vector
+//   Contains17(arr, size);               // pointer + size
+//   ...
+//
+// It's also kind of messy to have two separate arguments for what is
+// conceptually a single thing.
+//
+// Enter rtc::ArrayView<T>. It contains a T pointer (to an array it doesn't
+// own) and a count, and supports the basic things you'd expect, such as
+// indexing and iteration. It allows us to write our function like this:
+//
+//   bool Contains17(rtc::ArrayView<const int> arr) {
+//     for (auto e : arr) {
+//       if (e == 17)
+//         return true;
+//     }
+//     return false;
+//   }
+//
+// And even better, because a bunch of things will implicitly convert to
+// ArrayView, we can call it like this:
+//
+//   Contains17(arr);                             // C array
+//   Contains17(arr);                             // std::vector
+//   Contains17(rtc::ArrayView<int>(arr, size));  // pointer + size
+//   Contains17(nullptr);                         // nullptr -> empty ArrayView
+//   ...
+//
+// ArrayView<T> stores both a pointer and a size, but you may also use
+// ArrayView<T, N>, which has a size that's fixed at compile time (which means
+// it only has to store the pointer).
+//
+// One important point is that ArrayView<T> and ArrayView<const T> are
+// different types, which allow and don't allow mutation of the array elements,
+// respectively. The implicit conversions work just like you'd hope, so that
+// e.g. vector<int> will convert to either ArrayView<int> or ArrayView<const
+// int>, but const vector<int> will convert only to ArrayView<const int>.
+// (ArrayView itself can be the source type in such conversions, so
+// ArrayView<int> will convert to ArrayView<const int>.)
+//
+// Note: ArrayView is tiny (just a pointer and a count if variable-sized, just
+// a pointer if fix-sized) and trivially copyable, so it's probably cheaper to
+// pass it by value than by const reference.
+
+namespace array_view_internal {
+
+// Magic constant for indicating that the size of an ArrayView is variable
+// instead of fixed.
+enum : std::ptrdiff_t { kArrayViewVarSize = -4711 };
+
+// Base class for ArrayViews of fixed nonzero size.
+template <typename T, std::ptrdiff_t Size>
+class ArrayViewBase {
+  static_assert(Size > 0, "ArrayView size must be variable or non-negative");
+
+ public:
+  ArrayViewBase(T* data, size_t size) : data_(data) {}
+
+  static constexpr size_t size() { return Size; }
+  static constexpr bool empty() { return false; }
+  T* data() const { return data_; }
+
+ protected:
+  static constexpr bool fixed_size() { return true; }
+
+ private:
+  T* data_;
+};
+
+// Specialized base class for ArrayViews of fixed zero size.
+template <typename T>
+class ArrayViewBase<T, 0> {
+ public:
+  explicit ArrayViewBase(T* data, size_t size) {}
+
+  static constexpr size_t size() { return 0; }
+  static constexpr bool empty() { return true; }
+  T* data() const { return nullptr; }
+
+ protected:
+  static constexpr bool fixed_size() { return true; }
+};
+
+// Specialized base class for ArrayViews of variable size.
+template <typename T>
+class ArrayViewBase<T, array_view_internal::kArrayViewVarSize> {
+ public:
+  ArrayViewBase(T* data, size_t size)
+      : data_(size == 0 ? nullptr : data), size_(size) {}
+
+  size_t size() const { return size_; }
+  bool empty() const { return size_ == 0; }
+  T* data() const { return data_; }
+
+ protected:
+  static constexpr bool fixed_size() { return false; }
+
+ private:
+  T* data_;
+  size_t size_;
+};
+
+}  // namespace array_view_internal
+
+template <typename T,
+          std::ptrdiff_t Size = array_view_internal::kArrayViewVarSize>
+class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
+ public:
+  using value_type = T;
+  using reference = value_type&;
+  using const_reference = const value_type&;
+  using pointer = value_type*;
+  using const_pointer = const value_type*;
+  using const_iterator = const T*;
+
+  // Construct an ArrayView from a pointer and a length.
+  template <typename U>
+  ArrayView(U* data, size_t size)
+      : array_view_internal::ArrayViewBase<T, Size>::ArrayViewBase(data, size) {
+    RTC_DCHECK_EQ(size == 0 ? nullptr : data, this->data());
+    RTC_DCHECK_EQ(size, this->size());
+    RTC_DCHECK_EQ(!this->data(),
+                  this->size() == 0);  // data is null iff size == 0.
+  }
+
+  // Construct an empty ArrayView. Note that fixed-size ArrayViews of size > 0
+  // cannot be empty.
+  ArrayView() : ArrayView(nullptr, 0) {}
+  ArrayView(std::nullptr_t)  // NOLINT
+      : ArrayView() {}
+  ArrayView(std::nullptr_t, size_t size)
+      : ArrayView(static_cast<T*>(nullptr), size) {
+    static_assert(Size == 0 || Size == array_view_internal::kArrayViewVarSize,
+                  "");
+    RTC_DCHECK_EQ(0, size);
+  }
+
+  // Construct an ArrayView from a C-style array.
+  template <typename U, size_t N>
+  ArrayView(U (&array)[N])  // NOLINT
+      : ArrayView(array, N) {
+    static_assert(Size == N || Size == array_view_internal::kArrayViewVarSize,
+                  "Array size must match ArrayView size");
+  }
+
+  // (Only if size is fixed.) Construct a fixed size ArrayView<T, N> from a
+  // non-const std::array instance. For an ArrayView with variable size, the
+  // used ctor is ArrayView(U& u) instead.
+  template <typename U,
+            size_t N,
+            typename std::enable_if<
+                Size == static_cast<std::ptrdiff_t>(N)>::type* = nullptr>
+  ArrayView(std::array<U, N>& u)  // NOLINT
+      : ArrayView(u.data(), u.size()) {}
+
+  // (Only if size is fixed.) Construct a fixed size ArrayView<T, N> where T is
+  // const from a const(expr) std::array instance. For an ArrayView with
+  // variable size, the used ctor is ArrayView(U& u) instead.
+  template <typename U,
+            size_t N,
+            typename std::enable_if<
+                Size == static_cast<std::ptrdiff_t>(N)>::type* = nullptr>
+  ArrayView(const std::array<U, N>& u)  // NOLINT
+      : ArrayView(u.data(), u.size()) {}
+
+  // (Only if size is fixed.) Construct an ArrayView from any type U that has a
+  // static constexpr size() method whose return value is equal to Size, and a
+  // data() method whose return value converts implicitly to T*. In particular,
+  // this means we allow conversion from ArrayView<T, N> to ArrayView<const T,
+  // N>, but not the other way around. We also don't allow conversion from
+  // ArrayView<T> to ArrayView<T, N>, or from ArrayView<T, M> to ArrayView<T,
+  // N> when M != N.
+  template <
+      typename U,
+      typename std::enable_if<Size != array_view_internal::kArrayViewVarSize &&
+                              HasDataAndSize<U, T>::value>::type* = nullptr>
+  ArrayView(U& u)  // NOLINT
+      : ArrayView(u.data(), u.size()) {
+    static_assert(U::size() == Size, "Sizes must match exactly");
+  }
+  template <
+      typename U,
+      typename std::enable_if<Size != array_view_internal::kArrayViewVarSize &&
+                              HasDataAndSize<U, T>::value>::type* = nullptr>
+  ArrayView(const U& u)  // NOLINT(runtime/explicit)
+      : ArrayView(u.data(), u.size()) {
+    static_assert(U::size() == Size, "Sizes must match exactly");
+  }
+
+  // (Only if size is variable.) Construct an ArrayView from any type U that
+  // has a size() method whose return value converts implicitly to size_t, and
+  // a data() method whose return value converts implicitly to T*. In
+  // particular, this means we allow conversion from ArrayView<T> to
+  // ArrayView<const T>, but not the other way around. Other allowed
+  // conversions include
+  // ArrayView<T, N> to ArrayView<T> or ArrayView<const T>,
+  // std::vector<T> to ArrayView<T> or ArrayView<const T>,
+  // const std::vector<T> to ArrayView<const T>,
+  // rtc::Buffer to ArrayView<uint8_t> or ArrayView<const uint8_t>, and
+  // const rtc::Buffer to ArrayView<const uint8_t>.
+  template <
+      typename U,
+      typename std::enable_if<Size == array_view_internal::kArrayViewVarSize &&
+                              HasDataAndSize<U, T>::value>::type* = nullptr>
+  ArrayView(U& u)  // NOLINT
+      : ArrayView(u.data(), u.size()) {}
+  template <
+      typename U,
+      typename std::enable_if<Size == array_view_internal::kArrayViewVarSize &&
+                              HasDataAndSize<U, T>::value>::type* = nullptr>
+  ArrayView(const U& u)  // NOLINT(runtime/explicit)
+      : ArrayView(u.data(), u.size()) {}
+
+  // Indexing and iteration. These allow mutation even if the ArrayView is
+  // const, because the ArrayView doesn't own the array. (To prevent mutation,
+  // use a const element type.)
+  T& operator[](size_t idx) const {
+    RTC_DCHECK_LT(idx, this->size());
+    RTC_DCHECK(this->data());
+    return this->data()[idx];
+  }
+  T* begin() const { return this->data(); }
+  T* end() const { return this->data() + this->size(); }
+  const T* cbegin() const { return this->data(); }
+  const T* cend() const { return this->data() + this->size(); }
+  std::reverse_iterator<T*> rbegin() const {
+    return std::make_reverse_iterator(end());
+  }
+  std::reverse_iterator<T*> rend() const {
+    return std::make_reverse_iterator(begin());
+  }
+  std::reverse_iterator<const T*> crbegin() const {
+    return std::make_reverse_iterator(cend());
+  }
+  std::reverse_iterator<const T*> crend() const {
+    return std::make_reverse_iterator(cbegin());
+  }
+
+  ArrayView<T> subview(size_t offset, size_t size) const {
+    return offset < this->size()
+               ? ArrayView<T>(this->data() + offset,
+                              std::min(size, this->size() - offset))
+               : ArrayView<T>();
+  }
+  ArrayView<T> subview(size_t offset) const {
+    return subview(offset, this->size());
+  }
+};
+
+// Comparing two ArrayViews compares their (pointer,size) pairs; it does *not*
+// dereference the pointers.
+template <typename T, std::ptrdiff_t Size1, std::ptrdiff_t Size2>
+bool operator==(const ArrayView<T, Size1>& a, const ArrayView<T, Size2>& b) {
+  return a.data() == b.data() && a.size() == b.size();
+}
+template <typename T, std::ptrdiff_t Size1, std::ptrdiff_t Size2>
+bool operator!=(const ArrayView<T, Size1>& a, const ArrayView<T, Size2>& b) {
+  return !(a == b);
+}
+
+// Variable-size ArrayViews are the size of two pointers; fixed-size ArrayViews
+// are the size of one pointer. (And as a special case, fixed-size ArrayViews
+// of size 0 require no storage.)
+static_assert(sizeof(ArrayView<int>) == 2 * sizeof(int*), "");
+static_assert(sizeof(ArrayView<int, 17>) == sizeof(int*), "");
+static_assert(std::is_empty<ArrayView<int, 0>>::value, "");
+
+template <typename T>
+inline ArrayView<T> MakeArrayView(T* data, size_t size) {
+  return ArrayView<T>(data, size);
+}
+
+// Only for primitive types that have the same size and aligment.
+// Allow reinterpret cast of the array view to another primitive type of the
+// same size.
+// Template arguments order is (U, T, Size) to allow deduction of the template
+// arguments in client calls: reinterpret_array_view<target_type>(array_view).
+template <typename U, typename T, std::ptrdiff_t Size>
+inline ArrayView<U, Size> reinterpret_array_view(ArrayView<T, Size> view) {
+  static_assert(sizeof(U) == sizeof(T) && alignof(U) == alignof(T),
+                "ArrayView reinterpret_cast is only supported for casting "
+                "between views that represent the same chunk of memory.");
+  static_assert(
+      std::is_fundamental<T>::value && std::is_fundamental<U>::value,
+      "ArrayView reinterpret_cast is only supported for casting between "
+      "fundamental types.");
+  return ArrayView<U, Size>(reinterpret_cast<U*>(view.data()), view.size());
+}
+
+}  // namespace rtc
+
+#endif  // API_ARRAY_VIEW_H_

TMessagesProj/jni/voip/webrtc/api/array_view_unittest.cc

@@ -0,0 +1,630 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/array_view.h"
+
+#include <algorithm>
+#include <array>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/gunit.h"
+#include "test/gmock.h"
+
+namespace rtc {
+
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+
+template <typename T>
+size_t Call(ArrayView<T> av) {
+  return av.size();
+}
+
+template <typename T, size_t N>
+void CallFixed(ArrayView<T, N> av) {}
+
+}  // namespace
+
+TEST(ArrayViewDeathTest, TestConstructFromPtrAndArray) {
+  char arr[] = "Arrr!";
+  const char carr[] = "Carrr!";
+  EXPECT_EQ(6u, Call<const char>(arr));
+  EXPECT_EQ(7u, Call<const char>(carr));
+  EXPECT_EQ(6u, Call<char>(arr));
+  // Call<char>(carr);  // Compile error, because can't drop const.
+  // Call<int>(arr);  // Compile error, because incompatible types.
+  ArrayView<int*> x;
+  EXPECT_EQ(0u, x.size());
+  EXPECT_EQ(nullptr, x.data());
+  ArrayView<char> y = arr;
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<char, 6> yf = arr;
+  static_assert(yf.size() == 6, "");
+  EXPECT_EQ(arr, yf.data());
+  ArrayView<const char> z(arr + 1, 3);
+  EXPECT_EQ(3u, z.size());
+  EXPECT_EQ(arr + 1, z.data());
+  ArrayView<const char, 3> zf(arr + 1, 3);
+  static_assert(zf.size() == 3, "");
+  EXPECT_EQ(arr + 1, zf.data());
+  ArrayView<const char> w(arr, 2);
+  EXPECT_EQ(2u, w.size());
+  EXPECT_EQ(arr, w.data());
+  ArrayView<const char, 2> wf(arr, 2);
+  static_assert(wf.size() == 2, "");
+  EXPECT_EQ(arr, wf.data());
+  ArrayView<char> q(arr, 0);
+  EXPECT_EQ(0u, q.size());
+  EXPECT_EQ(nullptr, q.data());
+  ArrayView<char, 0> qf(arr, 0);
+  static_assert(qf.size() == 0, "");
+  EXPECT_EQ(nullptr, qf.data());
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+  // DCHECK error (nullptr with nonzero size).
+  EXPECT_DEATH(ArrayView<int>(static_cast<int*>(nullptr), 5), "");
+#endif
+  // These are compile errors, because incompatible types.
+  // ArrayView<int> m = arr;
+  // ArrayView<float> n(arr + 2, 2);
+}
+
+TEST(ArrayViewTest, TestCopyConstructorVariableLvalue) {
+  char arr[] = "Arrr!";
+  ArrayView<char> x = arr;
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<char> y = x;  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char> z = x;  // Copy non-const -> const.
+  EXPECT_EQ(6u, z.size());
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char> w = z;  // Copy const -> const.
+  EXPECT_EQ(6u, w.size());
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char> v = z;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyConstructorVariableRvalue) {
+  char arr[] = "Arrr!";
+  ArrayView<char> x = arr;
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<char> y = std::move(x);  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char> z = std::move(x);  // Copy non-const -> const.
+  EXPECT_EQ(6u, z.size());
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char> w = std::move(z);  // Copy const -> const.
+  EXPECT_EQ(6u, w.size());
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char> v = std::move(z);  // Error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyConstructorFixedLvalue) {
+  char arr[] = "Arrr!";
+  ArrayView<char, 6> x = arr;
+  static_assert(x.size() == 6, "");
+  EXPECT_EQ(arr, x.data());
+
+  // Copy fixed -> fixed.
+  ArrayView<char, 6> y = x;  // Copy non-const -> non-const.
+  static_assert(y.size() == 6, "");
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char, 6> z = x;  // Copy non-const -> const.
+  static_assert(z.size() == 6, "");
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char, 6> w = z;  // Copy const -> const.
+  static_assert(w.size() == 6, "");
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char, 6> v = z;  // Compile error, because can't drop const.
+
+  // Copy fixed -> variable.
+  ArrayView<char> yv = x;  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, yv.size());
+  EXPECT_EQ(arr, yv.data());
+  ArrayView<const char> zv = x;  // Copy non-const -> const.
+  EXPECT_EQ(6u, zv.size());
+  EXPECT_EQ(arr, zv.data());
+  ArrayView<const char> wv = z;  // Copy const -> const.
+  EXPECT_EQ(6u, wv.size());
+  EXPECT_EQ(arr, wv.data());
+  // ArrayView<char> vv = z;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyConstructorFixedRvalue) {
+  char arr[] = "Arrr!";
+  ArrayView<char, 6> x = arr;
+  static_assert(x.size() == 6, "");
+  EXPECT_EQ(arr, x.data());
+
+  // Copy fixed -> fixed.
+  ArrayView<char, 6> y = std::move(x);  // Copy non-const -> non-const.
+  static_assert(y.size() == 6, "");
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char, 6> z = std::move(x);  // Copy non-const -> const.
+  static_assert(z.size() == 6, "");
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char, 6> w = std::move(z);  // Copy const -> const.
+  static_assert(w.size() == 6, "");
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char, 6> v = std::move(z);  // Error, because can't drop const.
+
+  // Copy fixed -> variable.
+  ArrayView<char> yv = std::move(x);  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, yv.size());
+  EXPECT_EQ(arr, yv.data());
+  ArrayView<const char> zv = std::move(x);  // Copy non-const -> const.
+  EXPECT_EQ(6u, zv.size());
+  EXPECT_EQ(arr, zv.data());
+  ArrayView<const char> wv = std::move(z);  // Copy const -> const.
+  EXPECT_EQ(6u, wv.size());
+  EXPECT_EQ(arr, wv.data());
+  // ArrayView<char> vv = std::move(z);  // Error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyAssignmentVariableLvalue) {
+  char arr[] = "Arrr!";
+  ArrayView<char> x(arr);
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<char> y;
+  y = x;  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char> z;
+  z = x;  // Copy non-const -> const.
+  EXPECT_EQ(6u, z.size());
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char> w;
+  w = z;  // Copy const -> const.
+  EXPECT_EQ(6u, w.size());
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char> v;
+  // v = z;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyAssignmentVariableRvalue) {
+  char arr[] = "Arrr!";
+  ArrayView<char> x(arr);
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<char> y;
+  y = std::move(x);  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char> z;
+  z = std::move(x);  // Copy non-const -> const.
+  EXPECT_EQ(6u, z.size());
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char> w;
+  w = std::move(z);  // Copy const -> const.
+  EXPECT_EQ(6u, w.size());
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char> v;
+  // v = std::move(z);  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyAssignmentFixedLvalue) {
+  char arr[] = "Arrr!";
+  char init[] = "Init!";
+  ArrayView<char, 6> x(arr);
+  EXPECT_EQ(arr, x.data());
+
+  // Copy fixed -> fixed.
+  ArrayView<char, 6> y(init);
+  y = x;  // Copy non-const -> non-const.
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char, 6> z(init);
+  z = x;  // Copy non-const -> const.
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char, 6> w(init);
+  w = z;  // Copy const -> const.
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char, 6> v(init);
+  // v = z;  // Compile error, because can't drop const.
+
+  // Copy fixed -> variable.
+  ArrayView<char> yv;
+  yv = x;  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, yv.size());
+  EXPECT_EQ(arr, yv.data());
+  ArrayView<const char> zv;
+  zv = x;  // Copy non-const -> const.
+  EXPECT_EQ(6u, zv.size());
+  EXPECT_EQ(arr, zv.data());
+  ArrayView<const char> wv;
+  wv = z;  // Copy const -> const.
+  EXPECT_EQ(6u, wv.size());
+  EXPECT_EQ(arr, wv.data());
+  // ArrayView<char> v;
+  // v = z;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestCopyAssignmentFixedRvalue) {
+  char arr[] = "Arrr!";
+  char init[] = "Init!";
+  ArrayView<char, 6> x(arr);
+  EXPECT_EQ(arr, x.data());
+
+  // Copy fixed -> fixed.
+  ArrayView<char, 6> y(init);
+  y = std::move(x);  // Copy non-const -> non-const.
+  EXPECT_EQ(arr, y.data());
+  ArrayView<const char, 6> z(init);
+  z = std::move(x);  // Copy non-const -> const.
+  EXPECT_EQ(arr, z.data());
+  ArrayView<const char, 6> w(init);
+  w = std::move(z);  // Copy const -> const.
+  EXPECT_EQ(arr, w.data());
+  // ArrayView<char, 6> v(init);
+  // v = std::move(z);  // Compile error, because can't drop const.
+
+  // Copy fixed -> variable.
+  ArrayView<char> yv;
+  yv = std::move(x);  // Copy non-const -> non-const.
+  EXPECT_EQ(6u, yv.size());
+  EXPECT_EQ(arr, yv.data());
+  ArrayView<const char> zv;
+  zv = std::move(x);  // Copy non-const -> const.
+  EXPECT_EQ(6u, zv.size());
+  EXPECT_EQ(arr, zv.data());
+  ArrayView<const char> wv;
+  wv = std::move(z);  // Copy const -> const.
+  EXPECT_EQ(6u, wv.size());
+  EXPECT_EQ(arr, wv.data());
+  // ArrayView<char> v;
+  // v = std::move(z);  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestStdArray) {
+  EXPECT_EQ(4u, Call<const int>(std::array<int, 4>{1, 2, 3, 4}));
+  CallFixed<const int, 3>(std::array<int, 3>{2, 3, 4});
+  constexpr size_t size = 5;
+  std::array<float, size> arr{};
+  // Fixed size view.
+  rtc::ArrayView<float, size> arr_view_fixed(arr);
+  EXPECT_EQ(arr.data(), arr_view_fixed.data());
+  static_assert(size == arr_view_fixed.size(), "");
+  // Variable size view.
+  rtc::ArrayView<float> arr_view(arr);
+  EXPECT_EQ(arr.data(), arr_view.data());
+  EXPECT_EQ(size, arr_view.size());
+}
+
+TEST(ArrayViewTest, TestConstStdArray) {
+  constexpr size_t size = 5;
+
+  constexpr std::array<float, size> constexpr_arr{};
+  rtc::ArrayView<const float, size> constexpr_arr_view(constexpr_arr);
+  EXPECT_EQ(constexpr_arr.data(), constexpr_arr_view.data());
+  static_assert(constexpr_arr.size() == constexpr_arr_view.size(), "");
+
+  const std::array<float, size> const_arr{};
+  rtc::ArrayView<const float, size> const_arr_view(const_arr);
+  EXPECT_EQ(const_arr.data(), const_arr_view.data());
+  static_assert(const_arr.size() == const_arr_view.size(), "");
+
+  std::array<float, size> non_const_arr{};
+  rtc::ArrayView<const float, size> non_const_arr_view(non_const_arr);
+  EXPECT_EQ(non_const_arr.data(), non_const_arr_view.data());
+  static_assert(non_const_arr.size() == non_const_arr_view.size(), "");
+}
+
+TEST(ArrayViewTest, TestStdVector) {
+  EXPECT_EQ(3u, Call<const int>(std::vector<int>{4, 5, 6}));
+  std::vector<int> v;
+  v.push_back(3);
+  v.push_back(11);
+  EXPECT_EQ(2u, Call<const int>(v));
+  EXPECT_EQ(2u, Call<int>(v));
+  // Call<unsigned int>(v);  // Compile error, because incompatible types.
+  ArrayView<int> x = v;
+  EXPECT_EQ(2u, x.size());
+  EXPECT_EQ(v.data(), x.data());
+  ArrayView<const int> y;
+  y = v;
+  EXPECT_EQ(2u, y.size());
+  EXPECT_EQ(v.data(), y.data());
+  // ArrayView<double> d = v;  // Compile error, because incompatible types.
+  const std::vector<int> cv;
+  EXPECT_EQ(0u, Call<const int>(cv));
+  // Call<int>(cv);  // Compile error, because can't drop const.
+  ArrayView<const int> z = cv;
+  EXPECT_EQ(0u, z.size());
+  EXPECT_EQ(nullptr, z.data());
+  // ArrayView<int> w = cv;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestRtcBuffer) {
+  rtc::Buffer b = "so buffer";
+  EXPECT_EQ(10u, Call<const uint8_t>(b));
+  EXPECT_EQ(10u, Call<uint8_t>(b));
+  // Call<int8_t>(b);  // Compile error, because incompatible types.
+  ArrayView<uint8_t> x = b;
+  EXPECT_EQ(10u, x.size());
+  EXPECT_EQ(b.data(), x.data());
+  ArrayView<const uint8_t> y;
+  y = b;
+  EXPECT_EQ(10u, y.size());
+  EXPECT_EQ(b.data(), y.data());
+  // ArrayView<char> d = b;  // Compile error, because incompatible types.
+  const rtc::Buffer cb = "very const";
+  EXPECT_EQ(11u, Call<const uint8_t>(cb));
+  // Call<uint8_t>(cb);  // Compile error, because can't drop const.
+  ArrayView<const uint8_t> z = cb;
+  EXPECT_EQ(11u, z.size());
+  EXPECT_EQ(cb.data(), z.data());
+  // ArrayView<uint8_t> w = cb;  // Compile error, because can't drop const.
+}
+
+TEST(ArrayViewTest, TestSwapVariable) {
+  const char arr[] = "Arrr!";
+  const char aye[] = "Aye, Cap'n!";
+  ArrayView<const char> x(arr);
+  EXPECT_EQ(6u, x.size());
+  EXPECT_EQ(arr, x.data());
+  ArrayView<const char> y(aye);
+  EXPECT_EQ(12u, y.size());
+  EXPECT_EQ(aye, y.data());
+  using std::swap;
+  swap(x, y);
+  EXPECT_EQ(12u, x.size());
+  EXPECT_EQ(aye, x.data());
+  EXPECT_EQ(6u, y.size());
+  EXPECT_EQ(arr, y.data());
+  // ArrayView<char> z;
+  // swap(x, z);  // Compile error, because can't drop const.
+}
+
+TEST(FixArrayViewTest, TestSwapFixed) {
+  const char arr[] = "Arr!";
+  char aye[] = "Aye!";
+  ArrayView<const char, 5> x(arr);
+  EXPECT_EQ(arr, x.data());
+  ArrayView<const char, 5> y(aye);
+  EXPECT_EQ(aye, y.data());
+  using std::swap;
+  swap(x, y);
+  EXPECT_EQ(aye, x.data());
+  EXPECT_EQ(arr, y.data());
+  // ArrayView<char, 5> z(aye);
+  // swap(x, z);  // Compile error, because can't drop const.
+  // ArrayView<const char, 4> w(aye, 4);
+  // swap(x, w);  // Compile error, because different sizes.
+}
+
+TEST(ArrayViewDeathTest, TestIndexing) {
+  char arr[] = "abcdefg";
+  ArrayView<char> x(arr);
+  const ArrayView<char> y(arr);
+  ArrayView<const char, 8> z(arr);
+  EXPECT_EQ(8u, x.size());
+  EXPECT_EQ(8u, y.size());
+  EXPECT_EQ(8u, z.size());
+  EXPECT_EQ('b', x[1]);
+  EXPECT_EQ('c', y[2]);
+  EXPECT_EQ('d', z[3]);
+  x[3] = 'X';
+  y[2] = 'Y';
+  // z[1] = 'Z';  // Compile error, because z's element type is const char.
+  EXPECT_EQ('b', x[1]);
+  EXPECT_EQ('Y', y[2]);
+  EXPECT_EQ('X', z[3]);
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+  EXPECT_DEATH(z[8], "");  // DCHECK error (index out of bounds).
+#endif
+}
+
+TEST(ArrayViewTest, TestIterationEmpty) {
+  // Variable-size.
+  ArrayView<std::vector<std::vector<std::vector<std::string>>>> av;
+  EXPECT_EQ(av.begin(), av.end());
+  EXPECT_EQ(av.cbegin(), av.cend());
+  for (auto& e : av) {
+    EXPECT_TRUE(false);
+    EXPECT_EQ(42u, e.size());  // Dummy use of e to prevent unused var warning.
+  }
+
+  // Fixed-size.
+  ArrayView<std::vector<std::vector<std::vector<std::string>>>, 0> af;
+  EXPECT_EQ(af.begin(), af.end());
+  EXPECT_EQ(af.cbegin(), af.cend());
+  for (auto& e : af) {
+    EXPECT_TRUE(false);
+    EXPECT_EQ(42u, e.size());  // Dummy use of e to prevent unused var warning.
+  }
+}
+
+TEST(ArrayViewTest, TestReverseIterationEmpty) {
+  // Variable-size.
+  ArrayView<std::vector<std::vector<std::vector<std::string>>>> av;
+  EXPECT_EQ(av.rbegin(), av.rend());
+  EXPECT_EQ(av.crbegin(), av.crend());
+  EXPECT_TRUE(av.empty());
+
+  // Fixed-size.
+  ArrayView<std::vector<std::vector<std::vector<std::string>>>, 0> af;
+  EXPECT_EQ(af.begin(), af.end());
+  EXPECT_EQ(af.cbegin(), af.cend());
+  EXPECT_TRUE(af.empty());
+}
+
+TEST(ArrayViewTest, TestIterationVariable) {
+  char arr[] = "Arrr!";
+  ArrayView<char> av(arr);
+  EXPECT_EQ('A', *av.begin());
+  EXPECT_EQ('A', *av.cbegin());
+  EXPECT_EQ('\0', *(av.end() - 1));
+  EXPECT_EQ('\0', *(av.cend() - 1));
+  char i = 0;
+  for (auto& e : av) {
+    EXPECT_EQ(arr + i, &e);
+    e = 's' + i;
+    ++i;
+  }
+  i = 0;
+  for (auto& e : ArrayView<const char>(av)) {
+    EXPECT_EQ(arr + i, &e);
+    // e = 'q' + i;  // Compile error, because e is a const char&.
+    ++i;
+  }
+}
+
+TEST(ArrayViewTest, TestReverseIterationVariable) {
+  char arr[] = "Arrr!";
+  ArrayView<char> av(arr);
+  EXPECT_EQ('\0', *av.rbegin());
+  EXPECT_EQ('\0', *av.crbegin());
+  EXPECT_EQ('A', *(av.rend() - 1));
+  EXPECT_EQ('A', *(av.crend() - 1));
+
+  const char* cit = av.cend() - 1;
+  for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
+    EXPECT_EQ(*cit, *crit);
+  }
+
+  char* it = av.end() - 1;
+  for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
+    EXPECT_EQ(*it, *rit);
+  }
+}
+
+TEST(ArrayViewTest, TestIterationFixed) {
+  char arr[] = "Arrr!";
+  ArrayView<char, 6> av(arr);
+  EXPECT_EQ('A', *av.begin());
+  EXPECT_EQ('A', *av.cbegin());
+  EXPECT_EQ('\0', *(av.end() - 1));
+  EXPECT_EQ('\0', *(av.cend() - 1));
+  char i = 0;
+  for (auto& e : av) {
+    EXPECT_EQ(arr + i, &e);
+    e = 's' + i;
+    ++i;
+  }
+  i = 0;
+  for (auto& e : ArrayView<const char, 6>(av)) {
+    EXPECT_EQ(arr + i, &e);
+    // e = 'q' + i;  // Compile error, because e is a const char&.
+    ++i;
+  }
+}
+
+TEST(ArrayViewTest, TestReverseIterationFixed) {
+  char arr[] = "Arrr!";
+  ArrayView<char, 6> av(arr);
+  EXPECT_EQ('\0', *av.rbegin());
+  EXPECT_EQ('\0', *av.crbegin());
+  EXPECT_EQ('A', *(av.rend() - 1));
+  EXPECT_EQ('A', *(av.crend() - 1));
+
+  const char* cit = av.cend() - 1;
+  for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
+    EXPECT_EQ(*cit, *crit);
+  }
+
+  char* it = av.end() - 1;
+  for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
+    EXPECT_EQ(*it, *rit);
+  }
+}
+
+TEST(ArrayViewTest, TestEmpty) {
+  EXPECT_TRUE(ArrayView<int>().empty());
+  const int a[] = {1, 2, 3};
+  EXPECT_FALSE(ArrayView<const int>(a).empty());
+
+  static_assert(ArrayView<int, 0>::empty(), "");
+  static_assert(!ArrayView<int, 3>::empty(), "");
+}
+
+TEST(ArrayViewTest, TestCompare) {
+  int a[] = {1, 2, 3};
+  int b[] = {1, 2, 3};
+
+  EXPECT_EQ(ArrayView<int>(a), ArrayView<int>(a));
+  EXPECT_EQ((ArrayView<int, 3>(a)), (ArrayView<int, 3>(a)));
+  EXPECT_EQ(ArrayView<int>(a), (ArrayView<int, 3>(a)));
+  EXPECT_EQ(ArrayView<int>(), ArrayView<int>());
+  EXPECT_EQ(ArrayView<int>(), ArrayView<int>(a, 0));
+  EXPECT_EQ(ArrayView<int>(a, 0), ArrayView<int>(b, 0));
+  EXPECT_EQ((ArrayView<int, 0>(a, 0)), ArrayView<int>());
+
+  EXPECT_NE(ArrayView<int>(a), ArrayView<int>(b));
+  EXPECT_NE((ArrayView<int, 3>(a)), (ArrayView<int, 3>(b)));
+  EXPECT_NE((ArrayView<int, 3>(a)), ArrayView<int>(b));
+  EXPECT_NE(ArrayView<int>(a), ArrayView<int>());
+  EXPECT_NE(ArrayView<int>(a), ArrayView<int>(a, 2));
+  EXPECT_NE((ArrayView<int, 3>(a)), (ArrayView<int, 2>(a, 2)));
+}
+
+TEST(ArrayViewTest, TestSubViewVariable) {
+  int a[] = {1, 2, 3};
+  ArrayView<int> av(a);
+
+  EXPECT_EQ(av.subview(0), av);
+
+  EXPECT_THAT(av.subview(1), ElementsAre(2, 3));
+  EXPECT_THAT(av.subview(2), ElementsAre(3));
+  EXPECT_THAT(av.subview(3), IsEmpty());
+  EXPECT_THAT(av.subview(4), IsEmpty());
+
+  EXPECT_THAT(av.subview(1, 0), IsEmpty());
+  EXPECT_THAT(av.subview(1, 1), ElementsAre(2));
+  EXPECT_THAT(av.subview(1, 2), ElementsAre(2, 3));
+  EXPECT_THAT(av.subview(1, 3), ElementsAre(2, 3));
+}
+
+TEST(ArrayViewTest, TestSubViewFixed) {
+  int a[] = {1, 2, 3};
+  ArrayView<int, 3> av(a);
+
+  EXPECT_EQ(av.subview(0), av);
+
+  EXPECT_THAT(av.subview(1), ElementsAre(2, 3));
+  EXPECT_THAT(av.subview(2), ElementsAre(3));
+  EXPECT_THAT(av.subview(3), IsEmpty());
+  EXPECT_THAT(av.subview(4), IsEmpty());
+
+  EXPECT_THAT(av.subview(1, 0), IsEmpty());
+  EXPECT_THAT(av.subview(1, 1), ElementsAre(2));
+  EXPECT_THAT(av.subview(1, 2), ElementsAre(2, 3));
+  EXPECT_THAT(av.subview(1, 3), ElementsAre(2, 3));
+}
+
+TEST(ArrayViewTest, TestReinterpretCastFixedSize) {
+  uint8_t bytes[] = {1, 2, 3};
+  ArrayView<uint8_t, 3> uint8_av(bytes);
+  ArrayView<int8_t, 3> int8_av = reinterpret_array_view<int8_t>(uint8_av);
+  EXPECT_EQ(int8_av.size(), uint8_av.size());
+  EXPECT_EQ(int8_av[0], 1);
+  EXPECT_EQ(int8_av[1], 2);
+  EXPECT_EQ(int8_av[2], 3);
+}
+
+TEST(ArrayViewTest, TestReinterpretCastVariableSize) {
+  std::vector<int8_t> v = {1, 2, 3};
+  ArrayView<int8_t> int8_av(v);
+  ArrayView<uint8_t> uint8_av = reinterpret_array_view<uint8_t>(int8_av);
+  EXPECT_EQ(int8_av.size(), uint8_av.size());
+  EXPECT_EQ(uint8_av[0], 1);
+  EXPECT_EQ(uint8_av[1], 2);
+  EXPECT_EQ(uint8_av[2], 3);
+}
+}  // namespace rtc

TMessagesProj/jni/voip/webrtc/api/async_dns_resolver.h

@@ -0,0 +1,105 @@
+/*
+ *  Copyright 2021 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_ASYNC_DNS_RESOLVER_H_
+#define API_ASYNC_DNS_RESOLVER_H_
+
+#include <functional>
+#include <memory>
+
+#include "absl/functional/any_invocable.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// This interface defines the methods to resolve a hostname asynchronously.
+// The AsyncDnsResolverInterface class encapsulates a single name query.
+//
+// Usage:
+//   std::unique_ptr<AsyncDnsResolverInterface> resolver =
+//        factory->Create(address-to-be-resolved, [r = resolver.get()]() {
+//     if (r->result.GetResolvedAddress(AF_INET, &addr) {
+//       // success
+//     } else {
+//       // failure
+//       error = r->result().GetError();
+//     }
+//     // Release resolver.
+//     resolver_list.erase(std::remove_if(resolver_list.begin(),
+//     resolver_list.end(),
+//                         [](refptr) { refptr.get() == r; });
+//   });
+//   resolver_list.push_back(std::move(resolver));
+
+class AsyncDnsResolverResult {
+ public:
+  virtual ~AsyncDnsResolverResult() = default;
+  // Returns true iff the address from `Start` was successfully resolved.
+  // If the address was successfully resolved, sets `addr` to a copy of the
+  // address from `Start` with the IP address set to the top most resolved
+  // address of `family` (`addr` will have both hostname and the resolved ip).
+  virtual bool GetResolvedAddress(int family,
+                                  rtc::SocketAddress* addr) const = 0;
+  // Returns error from resolver.
+  virtual int GetError() const = 0;
+};
+
+// The API for a single name query.
+// The constructor, destructor and all functions must be called from
+// the same sequence, and the callback will also be called on that sequence.
+// The class guarantees that the callback will not be called if the
+// resolver's destructor has been called.
+class RTC_EXPORT AsyncDnsResolverInterface {
+ public:
+  virtual ~AsyncDnsResolverInterface() = default;
+
+  // Start address resolution of the hostname in `addr`.
+  virtual void Start(const rtc::SocketAddress& addr,
+                     absl::AnyInvocable<void()> callback) = 0;
+  // Start address resolution of the hostname in `addr` matching `family`.
+  virtual void Start(const rtc::SocketAddress& addr,
+                     int family,
+                     absl::AnyInvocable<void()> callback) = 0;
+  virtual const AsyncDnsResolverResult& result() const = 0;
+};
+
+// An abstract factory for creating AsyncDnsResolverInterfaces. This allows
+// client applications to provide WebRTC with their own mechanism for
+// performing DNS resolution.
+class AsyncDnsResolverFactoryInterface {
+ public:
+  virtual ~AsyncDnsResolverFactoryInterface() = default;
+
+  // Creates an AsyncDnsResolver and starts resolving the name. The callback
+  // will be called when resolution is finished.
+  // The callback will be called on the sequence that the caller runs on.
+  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAndResolve(
+      const rtc::SocketAddress& addr,
+      absl::AnyInvocable<void()> callback) = 0;
+  // Creates an AsyncDnsResolver and starts resolving the name to an address
+  // matching the specified family. The callback will be called when resolution
+  // is finished. The callback will be called on the sequence that the caller
+  // runs on.
+  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAndResolve(
+      const rtc::SocketAddress& addr,
+      int family,
+      absl::AnyInvocable<void()> callback) = 0;
+  // Creates an AsyncDnsResolver and does not start it.
+  // For backwards compatibility, will be deprecated and removed.
+  // One has to do a separate Start() call on the
+  // resolver to start name resolution.
+  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> Create() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_ASYNC_DNS_RESOLVER_H_

TMessagesProj/jni/voip/webrtc/api/audio/BUILD.gn

@@ -0,0 +1,93 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("audio_frame_api") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_frame.cc",
+    "audio_frame.h",
+    "channel_layout.cc",
+    "channel_layout.h",
+  ]
+
+  deps = [
+    "..:rtp_packet_info",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:timeutils",
+  ]
+}
+
+rtc_source_set("audio_frame_processor") {
+  visibility = [ "*" ]
+  sources = [ "audio_frame_processor.h" ]
+}
+
+rtc_source_set("audio_mixer_api") {
+  visibility = [ "*" ]
+  sources = [ "audio_mixer.h" ]
+
+  deps = [
+    ":audio_frame_api",
+    "..:make_ref_counted",
+    "../../rtc_base:refcount",
+  ]
+}
+
+rtc_library("aec3_config") {
+  visibility = [ "*" ]
+  sources = [
+    "echo_canceller3_config.cc",
+    "echo_canceller3_config.h",
+  ]
+  deps = [
+    "../../rtc_base:checks",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base/system:rtc_export",
+  ]
+}
+
+rtc_library("aec3_factory") {
+  visibility = [ "*" ]
+  configs += [ "../../modules/audio_processing:apm_debug_dump" ]
+  sources = [
+    "echo_canceller3_factory.cc",
+    "echo_canceller3_factory.h",
+  ]
+
+  deps = [
+    ":aec3_config",
+    ":echo_control",
+    "../../modules/audio_processing/aec3",
+    "../../rtc_base/system:rtc_export",
+  ]
+}
+
+rtc_source_set("echo_control") {
+  visibility = [ "*" ]
+  sources = [ "echo_control.h" ]
+  deps = [ "../../rtc_base:checks" ]
+}
+
+rtc_source_set("echo_detector_creator") {
+  visibility = [ "*" ]
+  allow_poison = [ "default_echo_detector" ]
+  sources = [
+    "echo_detector_creator.cc",
+    "echo_detector_creator.h",
+  ]
+  deps = [
+    "..:make_ref_counted",
+    "../../api:scoped_refptr",
+    "../../modules/audio_processing:api",
+    "../../modules/audio_processing:residual_echo_detector",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio/OWNERS

@@ -0,0 +1,2 @@
+gustaf@webrtc.org
+peah@webrtc.org

TMessagesProj/jni/voip/webrtc/api/audio/audio_frame.cc

@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio/audio_frame.h"
+
+#include <string.h>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+AudioFrame::AudioFrame() {
+  // Visual Studio doesn't like this in the class definition.
+  static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
+}
+
+void AudioFrame::Reset() {
+  ResetWithoutMuting();
+  muted_ = true;
+}
+
+void AudioFrame::ResetWithoutMuting() {
+  // TODO(wu): Zero is a valid value for `timestamp_`. We should initialize
+  // to an invalid value, or add a new member to indicate invalidity.
+  timestamp_ = 0;
+  elapsed_time_ms_ = -1;
+  ntp_time_ms_ = -1;
+  samples_per_channel_ = 0;
+  sample_rate_hz_ = 0;
+  num_channels_ = 0;
+  channel_layout_ = CHANNEL_LAYOUT_NONE;
+  speech_type_ = kUndefined;
+  vad_activity_ = kVadUnknown;
+  profile_timestamp_ms_ = 0;
+  packet_infos_ = RtpPacketInfos();
+  absolute_capture_timestamp_ms_ = absl::nullopt;
+}
+
+void AudioFrame::UpdateFrame(uint32_t timestamp,
+                             const int16_t* data,
+                             size_t samples_per_channel,
+                             int sample_rate_hz,
+                             SpeechType speech_type,
+                             VADActivity vad_activity,
+                             size_t num_channels) {
+  timestamp_ = timestamp;
+  samples_per_channel_ = samples_per_channel;
+  sample_rate_hz_ = sample_rate_hz;
+  speech_type_ = speech_type;
+  vad_activity_ = vad_activity;
+  num_channels_ = num_channels;
+  channel_layout_ = GuessChannelLayout(num_channels);
+  if (channel_layout_ != CHANNEL_LAYOUT_UNSUPPORTED) {
+    RTC_DCHECK_EQ(num_channels, ChannelLayoutToChannelCount(channel_layout_));
+  }
+
+  const size_t length = samples_per_channel * num_channels;
+  RTC_CHECK_LE(length, kMaxDataSizeSamples);
+  if (data != nullptr) {
+    memcpy(data_, data, sizeof(int16_t) * length);
+    muted_ = false;
+  } else {
+    muted_ = true;
+  }
+}
+
+void AudioFrame::CopyFrom(const AudioFrame& src) {
+  if (this == &src)
+    return;
+
+  timestamp_ = src.timestamp_;
+  elapsed_time_ms_ = src.elapsed_time_ms_;
+  ntp_time_ms_ = src.ntp_time_ms_;
+  packet_infos_ = src.packet_infos_;
+  muted_ = src.muted();
+  samples_per_channel_ = src.samples_per_channel_;
+  sample_rate_hz_ = src.sample_rate_hz_;
+  speech_type_ = src.speech_type_;
+  vad_activity_ = src.vad_activity_;
+  num_channels_ = src.num_channels_;
+  channel_layout_ = src.channel_layout_;
+  absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms();
+
+  const size_t length = samples_per_channel_ * num_channels_;
+  RTC_CHECK_LE(length, kMaxDataSizeSamples);
+  if (!src.muted()) {
+    memcpy(data_, src.data(), sizeof(int16_t) * length);
+    muted_ = false;
+  }
+}
+
+void AudioFrame::UpdateProfileTimeStamp() {
+  profile_timestamp_ms_ = rtc::TimeMillis();
+}
+
+int64_t AudioFrame::ElapsedProfileTimeMs() const {
+  if (profile_timestamp_ms_ == 0) {
+    // Profiling has not been activated.
+    return -1;
+  }
+  return rtc::TimeSince(profile_timestamp_ms_);
+}
+
+const int16_t* AudioFrame::data() const {
+  return muted_ ? empty_data() : data_;
+}
+
+// TODO(henrik.lundin) Can we skip zeroing the buffer?
+// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
+int16_t* AudioFrame::mutable_data() {
+  if (muted_) {
+    memset(data_, 0, kMaxDataSizeBytes);
+    muted_ = false;
+  }
+  return data_;
+}
+
+void AudioFrame::Mute() {
+  muted_ = true;
+}
+
+bool AudioFrame::muted() const {
+  return muted_;
+}
+
+// static
+const int16_t* AudioFrame::empty_data() {
+  static int16_t* null_data = new int16_t[kMaxDataSizeSamples]();
+  return &null_data[0];
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio/audio_frame.h

@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_AUDIO_FRAME_H_
+#define API_AUDIO_AUDIO_FRAME_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/audio/channel_layout.h"
+#include "api/rtp_packet_infos.h"
+
+namespace webrtc {
+
+/* This class holds up to 120 ms of super-wideband (32 kHz) stereo audio. It
+ * allows for adding and subtracting frames while keeping track of the resulting
+ * states.
+ *
+ * Notes
+ * - This is a de-facto api, not designed for external use. The AudioFrame class
+ *   is in need of overhaul or even replacement, and anyone depending on it
+ *   should be prepared for that.
+ * - The total number of samples is samples_per_channel_ * num_channels_.
+ * - Stereo data is interleaved starting with the left channel.
+ */
+class AudioFrame {
+ public:
+  // Using constexpr here causes linker errors unless the variable also has an
+  // out-of-class definition, which is impractical in this header-only class.
+  // (This makes no sense because it compiles as an enum value, which we most
+  // certainly cannot take the address of, just fine.) C++17 introduces inline
+  // variables which should allow us to switch to constexpr and keep this a
+  // header-only class.
+  enum : size_t {
+    // Stereo, 32 kHz, 120 ms (2 * 32 * 120)
+    // Stereo, 192 kHz, 20 ms (2 * 192 * 20)
+    kMaxDataSizeSamples = 7680,
+    kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
+  };
+
+  enum VADActivity { kVadActive = 0, kVadPassive = 1, kVadUnknown = 2 };
+  enum SpeechType {
+    kNormalSpeech = 0,
+    kPLC = 1,
+    kCNG = 2,
+    kPLCCNG = 3,
+    kCodecPLC = 5,
+    kUndefined = 4
+  };
+
+  AudioFrame();
+
+  AudioFrame(const AudioFrame&) = delete;
+  AudioFrame& operator=(const AudioFrame&) = delete;
+
+  // Resets all members to their default state.
+  void Reset();
+  // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
+  // the buffer to be zeroed on the next call to mutable_data(). Callers
+  // intending to write to the buffer immediately after Reset() can instead use
+  // ResetWithoutMuting() to skip this wasteful zeroing.
+  void ResetWithoutMuting();
+
+  void UpdateFrame(uint32_t timestamp,
+                   const int16_t* data,
+                   size_t samples_per_channel,
+                   int sample_rate_hz,
+                   SpeechType speech_type,
+                   VADActivity vad_activity,
+                   size_t num_channels = 1);
+
+  void CopyFrom(const AudioFrame& src);
+
+  // Sets a wall-time clock timestamp in milliseconds to be used for profiling
+  // of time between two points in the audio chain.
+  // Example:
+  //   t0: UpdateProfileTimeStamp()
+  //   t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
+  void UpdateProfileTimeStamp();
+  // Returns the time difference between now and when UpdateProfileTimeStamp()
+  // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
+  // called.
+  int64_t ElapsedProfileTimeMs() const;
+
+  // data() returns a zeroed static buffer if the frame is muted.
+  // mutable_frame() always returns a non-static buffer; the first call to
+  // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
+  const int16_t* data() const;
+  int16_t* mutable_data();
+
+  // Prefer to mute frames using AudioFrameOperations::Mute.
+  void Mute();
+  // Frame is muted by default.
+  bool muted() const;
+
+  size_t max_16bit_samples() const { return kMaxDataSizeSamples; }
+  size_t samples_per_channel() const { return samples_per_channel_; }
+  size_t num_channels() const { return num_channels_; }
+  ChannelLayout channel_layout() const { return channel_layout_; }
+  int sample_rate_hz() const { return sample_rate_hz_; }
+
+  void set_absolute_capture_timestamp_ms(
+      int64_t absolute_capture_time_stamp_ms) {
+    absolute_capture_timestamp_ms_ = absolute_capture_time_stamp_ms;
+  }
+
+  absl::optional<int64_t> absolute_capture_timestamp_ms() const {
+    return absolute_capture_timestamp_ms_;
+  }
+
+  // RTP timestamp of the first sample in the AudioFrame.
+  uint32_t timestamp_ = 0;
+  // Time since the first frame in milliseconds.
+  // -1 represents an uninitialized value.
+  int64_t elapsed_time_ms_ = -1;
+  // NTP time of the estimated capture time in local timebase in milliseconds.
+  // -1 represents an uninitialized value.
+  int64_t ntp_time_ms_ = -1;
+  size_t samples_per_channel_ = 0;
+  int sample_rate_hz_ = 0;
+  size_t num_channels_ = 0;
+  ChannelLayout channel_layout_ = CHANNEL_LAYOUT_NONE;
+  SpeechType speech_type_ = kUndefined;
+  VADActivity vad_activity_ = kVadUnknown;
+  // Monotonically increasing timestamp intended for profiling of audio frames.
+  // Typically used for measuring elapsed time between two different points in
+  // the audio path. No lock is used to save resources and we are thread safe
+  // by design.
+  // TODO(nisse@webrtc.org): consider using absl::optional.
+  int64_t profile_timestamp_ms_ = 0;
+
+  // Information about packets used to assemble this audio frame. This is needed
+  // by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's
+  // MediaStreamTrack, in order to implement getContributingSources(). See:
+  // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
+  //
+  // TODO(bugs.webrtc.org/10757):
+  //   Note that this information might not be fully accurate since we currently
+  //   don't have a proper way to track it across the audio sync buffer. The
+  //   sync buffer is the small sample-holding buffer located after the audio
+  //   decoder and before where samples are assembled into output frames.
+  //
+  // `RtpPacketInfos` may also be empty if the audio samples did not come from
+  // RTP packets. E.g. if the audio were locally generated by packet loss
+  // concealment, comfort noise generation, etc.
+  RtpPacketInfos packet_infos_;
+
+ private:
+  // A permanently zeroed out buffer to represent muted frames. This is a
+  // header-only class, so the only way to avoid creating a separate empty
+  // buffer per translation unit is to wrap a static in an inline function.
+  static const int16_t* empty_data();
+
+  int16_t data_[kMaxDataSizeSamples];
+  bool muted_ = true;
+
+  // Absolute capture timestamp when this audio frame was originally captured.
+  // This is only valid for audio frames captured on this machine. The absolute
+  // capture timestamp of a received frame is found in `packet_infos_`.
+  // This timestamp MUST be based on the same clock as rtc::TimeMillis().
+  absl::optional<int64_t> absolute_capture_timestamp_ms_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_AUDIO_FRAME_H_

TMessagesProj/jni/voip/webrtc/api/audio/audio_frame_processor.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_AUDIO_FRAME_PROCESSOR_H_
+#define API_AUDIO_AUDIO_FRAME_PROCESSOR_H_
+
+#include <functional>
+#include <memory>
+
+namespace webrtc {
+
+class AudioFrame;
+
+// If passed into PeerConnectionFactory, will be used for additional
+// processing of captured audio frames, performed before encoding.
+// Implementations must be thread-safe.
+class AudioFrameProcessor {
+ public:
+  using OnAudioFrameCallback = std::function<void(std::unique_ptr<AudioFrame>)>;
+  virtual ~AudioFrameProcessor() = default;
+
+  // Processes the frame received from WebRTC, is called by WebRTC off the
+  // realtime audio capturing path. AudioFrameProcessor must reply with
+  // processed frames by calling `sink_callback` if it was provided in SetSink()
+  // call. `sink_callback` can be called in the context of Process().
+  virtual void Process(std::unique_ptr<AudioFrame> frame) = 0;
+
+  // Atomically replaces the current sink with the new one. Before the
+  // first call to this function, or if the provided `sink_callback` is nullptr,
+  // processed frames are simply discarded.
+  virtual void SetSink(OnAudioFrameCallback sink_callback) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_AUDIO_FRAME_PROCESSOR_H_

TMessagesProj/jni/voip/webrtc/api/audio/audio_mixer.h

@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_AUDIO_MIXER_H_
+#define API_AUDIO_AUDIO_MIXER_H_
+
+#include <memory>
+
+#include "api/audio/audio_frame.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle.
+class AudioMixer : public rtc::RefCountInterface {
+ public:
+  // A callback class that all mixer participants must inherit from/implement.
+  class Source {
+   public:
+    enum class AudioFrameInfo {
+      kNormal,  // The samples in audio_frame are valid and should be used.
+      kMuted,   // The samples in audio_frame should not be used, but
+                // should be implicitly interpreted as zero. Other
+                // fields in audio_frame may be read and should
+                // contain meaningful values.
+      kError,   // The audio_frame will not be used.
+    };
+
+    // Overwrites `audio_frame`. The data_ field is overwritten with
+    // 10 ms of new audio (either 1 or 2 interleaved channels) at
+    // `sample_rate_hz`. All fields in `audio_frame` must be updated.
+    virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                                 AudioFrame* audio_frame) = 0;
+
+    // A way for a mixer implementation to distinguish participants.
+    virtual int Ssrc() const = 0;
+
+    // A way for this source to say that GetAudioFrameWithInfo called
+    // with this sample rate or higher will not cause quality loss.
+    virtual int PreferredSampleRate() const = 0;
+
+    virtual ~Source() {}
+  };
+
+  // Returns true if adding was successful. A source is never added
+  // twice. Addition and removal can happen on different threads.
+  virtual bool AddSource(Source* audio_source) = 0;
+
+  // Removal is never attempted if a source has not been successfully
+  // added to the mixer.
+  virtual void RemoveSource(Source* audio_source) = 0;
+
+  // Performs mixing by asking registered audio sources for audio. The
+  // mixed result is placed in the provided AudioFrame. This method
+  // will only be called from a single thread. The channels argument
+  // specifies the number of channels of the mix result. The mixer
+  // should mix at a rate that doesn't cause quality loss of the
+  // sources' audio. The mixing rate is one of the rates listed in
+  // AudioProcessing::NativeRate. All fields in
+  // `audio_frame_for_mixing` must be updated.
+  virtual void Mix(size_t number_of_channels,
+                   AudioFrame* audio_frame_for_mixing) = 0;
+
+ protected:
+  // Since the mixer is reference counted, the destructor may be
+  // called from any thread.
+  ~AudioMixer() override {}
+};
+}  // namespace webrtc
+
+#endif  // API_AUDIO_AUDIO_MIXER_H_

TMessagesProj/jni/voip/webrtc/api/audio/channel_layout.cc

@@ -0,0 +1,282 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio/channel_layout.h"
+
+#include <stddef.h>
+
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static const int kLayoutToChannels[] = {
+    0,  // CHANNEL_LAYOUT_NONE
+    0,  // CHANNEL_LAYOUT_UNSUPPORTED
+    1,  // CHANNEL_LAYOUT_MONO
+    2,  // CHANNEL_LAYOUT_STEREO
+    3,  // CHANNEL_LAYOUT_2_1
+    3,  // CHANNEL_LAYOUT_SURROUND
+    4,  // CHANNEL_LAYOUT_4_0
+    4,  // CHANNEL_LAYOUT_2_2
+    4,  // CHANNEL_LAYOUT_QUAD
+    5,  // CHANNEL_LAYOUT_5_0
+    6,  // CHANNEL_LAYOUT_5_1
+    5,  // CHANNEL_LAYOUT_5_0_BACK
+    6,  // CHANNEL_LAYOUT_5_1_BACK
+    7,  // CHANNEL_LAYOUT_7_0
+    8,  // CHANNEL_LAYOUT_7_1
+    8,  // CHANNEL_LAYOUT_7_1_WIDE
+    2,  // CHANNEL_LAYOUT_STEREO_DOWNMIX
+    3,  // CHANNEL_LAYOUT_2POINT1
+    4,  // CHANNEL_LAYOUT_3_1
+    5,  // CHANNEL_LAYOUT_4_1
+    6,  // CHANNEL_LAYOUT_6_0
+    6,  // CHANNEL_LAYOUT_6_0_FRONT
+    6,  // CHANNEL_LAYOUT_HEXAGONAL
+    7,  // CHANNEL_LAYOUT_6_1
+    7,  // CHANNEL_LAYOUT_6_1_BACK
+    7,  // CHANNEL_LAYOUT_6_1_FRONT
+    7,  // CHANNEL_LAYOUT_7_0_FRONT
+    8,  // CHANNEL_LAYOUT_7_1_WIDE_BACK
+    8,  // CHANNEL_LAYOUT_OCTAGONAL
+    0,  // CHANNEL_LAYOUT_DISCRETE
+    3,  // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC
+    5,  // CHANNEL_LAYOUT_4_1_QUAD_SIDE
+    0,  // CHANNEL_LAYOUT_BITSTREAM
+};
+
+// The channel orderings for each layout as specified by FFmpeg. Each value
+// represents the index of each channel in each layout.  Values of -1 mean the
+// channel at that index is not used for that layout. For example, the left side
+// surround sound channel in FFmpeg's 5.1 layout is in the 5th position (because
+// the order is L, R, C, LFE, LS, RS), so
+// kChannelOrderings[CHANNEL_LAYOUT_5_1][SIDE_LEFT] = 4;
+static const int kChannelOrderings[CHANNEL_LAYOUT_MAX + 1][CHANNELS_MAX + 1] = {
+    // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR
+
+    // CHANNEL_LAYOUT_NONE
+    {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_UNSUPPORTED
+    {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_MONO
+    {-1, -1, 0, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_STEREO
+    {0, 1, -1, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_2_1
+    {0, 1, -1, -1, -1, -1, -1, -1, 2, -1, -1},
+
+    // CHANNEL_LAYOUT_SURROUND
+    {0, 1, 2, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_4_0
+    {0, 1, 2, -1, -1, -1, -1, -1, 3, -1, -1},
+
+    // CHANNEL_LAYOUT_2_2
+    {0, 1, -1, -1, -1, -1, -1, -1, -1, 2, 3},
+
+    // CHANNEL_LAYOUT_QUAD
+    {0, 1, -1, -1, 2, 3, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_5_0
+    {0, 1, 2, -1, -1, -1, -1, -1, -1, 3, 4},
+
+    // CHANNEL_LAYOUT_5_1
+    {0, 1, 2, 3, -1, -1, -1, -1, -1, 4, 5},
+
+    // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR
+
+    // CHANNEL_LAYOUT_5_0_BACK
+    {0, 1, 2, -1, 3, 4, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_5_1_BACK
+    {0, 1, 2, 3, 4, 5, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_7_0
+    {0, 1, 2, -1, 5, 6, -1, -1, -1, 3, 4},
+
+    // CHANNEL_LAYOUT_7_1
+    {0, 1, 2, 3, 6, 7, -1, -1, -1, 4, 5},
+
+    // CHANNEL_LAYOUT_7_1_WIDE
+    {0, 1, 2, 3, -1, -1, 6, 7, -1, 4, 5},
+
+    // CHANNEL_LAYOUT_STEREO_DOWNMIX
+    {0, 1, -1, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_2POINT1
+    {0, 1, -1, 2, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_3_1
+    {0, 1, 2, 3, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_4_1
+    {0, 1, 2, 4, -1, -1, -1, -1, 3, -1, -1},
+
+    // CHANNEL_LAYOUT_6_0
+    {0, 1, 2, -1, -1, -1, -1, -1, 5, 3, 4},
+
+    // CHANNEL_LAYOUT_6_0_FRONT
+    {0, 1, -1, -1, -1, -1, 4, 5, -1, 2, 3},
+
+    // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR
+
+    // CHANNEL_LAYOUT_HEXAGONAL
+    {0, 1, 2, -1, 3, 4, -1, -1, 5, -1, -1},
+
+    // CHANNEL_LAYOUT_6_1
+    {0, 1, 2, 3, -1, -1, -1, -1, 6, 4, 5},
+
+    // CHANNEL_LAYOUT_6_1_BACK
+    {0, 1, 2, 3, 4, 5, -1, -1, 6, -1, -1},
+
+    // CHANNEL_LAYOUT_6_1_FRONT
+    {0, 1, -1, 6, -1, -1, 4, 5, -1, 2, 3},
+
+    // CHANNEL_LAYOUT_7_0_FRONT
+    {0, 1, 2, -1, -1, -1, 5, 6, -1, 3, 4},
+
+    // CHANNEL_LAYOUT_7_1_WIDE_BACK
+    {0, 1, 2, 3, 4, 5, 6, 7, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_OCTAGONAL
+    {0, 1, 2, -1, 5, 6, -1, -1, 7, 3, 4},
+
+    // CHANNEL_LAYOUT_DISCRETE
+    {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC
+    {0, 1, 2, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // CHANNEL_LAYOUT_4_1_QUAD_SIDE
+    {0, 1, -1, 4, -1, -1, -1, -1, -1, 2, 3},
+
+    // CHANNEL_LAYOUT_BITSTREAM
+    {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1},
+
+    // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR
+};
+
+int ChannelLayoutToChannelCount(ChannelLayout layout) {
+  RTC_DCHECK_LT(static_cast<size_t>(layout), arraysize(kLayoutToChannels));
+  RTC_DCHECK_LE(kLayoutToChannels[layout], kMaxConcurrentChannels);
+  return kLayoutToChannels[layout];
+}
+
+// Converts a channel count into a channel layout.
+ChannelLayout GuessChannelLayout(int channels) {
+  switch (channels) {
+    case 1:
+      return CHANNEL_LAYOUT_MONO;
+    case 2:
+      return CHANNEL_LAYOUT_STEREO;
+    case 3:
+      return CHANNEL_LAYOUT_SURROUND;
+    case 4:
+      return CHANNEL_LAYOUT_QUAD;
+    case 5:
+      return CHANNEL_LAYOUT_5_0;
+    case 6:
+      return CHANNEL_LAYOUT_5_1;
+    case 7:
+      return CHANNEL_LAYOUT_6_1;
+    case 8:
+      return CHANNEL_LAYOUT_7_1;
+    default:
+      RTC_DLOG(LS_WARNING) << "Unsupported channel count: " << channels;
+  }
+  return CHANNEL_LAYOUT_UNSUPPORTED;
+}
+
+int ChannelOrder(ChannelLayout layout, Channels channel) {
+  RTC_DCHECK_LT(static_cast<size_t>(layout), arraysize(kChannelOrderings));
+  RTC_DCHECK_LT(static_cast<size_t>(channel), arraysize(kChannelOrderings[0]));
+  return kChannelOrderings[layout][channel];
+}
+
+const char* ChannelLayoutToString(ChannelLayout layout) {
+  switch (layout) {
+    case CHANNEL_LAYOUT_NONE:
+      return "NONE";
+    case CHANNEL_LAYOUT_UNSUPPORTED:
+      return "UNSUPPORTED";
+    case CHANNEL_LAYOUT_MONO:
+      return "MONO";
+    case CHANNEL_LAYOUT_STEREO:
+      return "STEREO";
+    case CHANNEL_LAYOUT_2_1:
+      return "2.1";
+    case CHANNEL_LAYOUT_SURROUND:
+      return "SURROUND";
+    case CHANNEL_LAYOUT_4_0:
+      return "4.0";
+    case CHANNEL_LAYOUT_2_2:
+      return "QUAD_SIDE";
+    case CHANNEL_LAYOUT_QUAD:
+      return "QUAD";
+    case CHANNEL_LAYOUT_5_0:
+      return "5.0";
+    case CHANNEL_LAYOUT_5_1:
+      return "5.1";
+    case CHANNEL_LAYOUT_5_0_BACK:
+      return "5.0_BACK";
+    case CHANNEL_LAYOUT_5_1_BACK:
+      return "5.1_BACK";
+    case CHANNEL_LAYOUT_7_0:
+      return "7.0";
+    case CHANNEL_LAYOUT_7_1:
+      return "7.1";
+    case CHANNEL_LAYOUT_7_1_WIDE:
+      return "7.1_WIDE";
+    case CHANNEL_LAYOUT_STEREO_DOWNMIX:
+      return "STEREO_DOWNMIX";
+    case CHANNEL_LAYOUT_2POINT1:
+      return "2POINT1";
+    case CHANNEL_LAYOUT_3_1:
+      return "3.1";
+    case CHANNEL_LAYOUT_4_1:
+      return "4.1";
+    case CHANNEL_LAYOUT_6_0:
+      return "6.0";
+    case CHANNEL_LAYOUT_6_0_FRONT:
+      return "6.0_FRONT";
+    case CHANNEL_LAYOUT_HEXAGONAL:
+      return "HEXAGONAL";
+    case CHANNEL_LAYOUT_6_1:
+      return "6.1";
+    case CHANNEL_LAYOUT_6_1_BACK:
+      return "6.1_BACK";
+    case CHANNEL_LAYOUT_6_1_FRONT:
+      return "6.1_FRONT";
+    case CHANNEL_LAYOUT_7_0_FRONT:
+      return "7.0_FRONT";
+    case CHANNEL_LAYOUT_7_1_WIDE_BACK:
+      return "7.1_WIDE_BACK";
+    case CHANNEL_LAYOUT_OCTAGONAL:
+      return "OCTAGONAL";
+    case CHANNEL_LAYOUT_DISCRETE:
+      return "DISCRETE";
+    case CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC:
+      return "STEREO_AND_KEYBOARD_MIC";
+    case CHANNEL_LAYOUT_4_1_QUAD_SIDE:
+      return "4.1_QUAD_SIDE";
+    case CHANNEL_LAYOUT_BITSTREAM:
+      return "BITSTREAM";
+  }
+  RTC_DCHECK_NOTREACHED() << "Invalid channel layout provided: " << layout;
+  return "";
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio/channel_layout.h

@@ -0,0 +1,165 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CHANNEL_LAYOUT_H_
+#define API_AUDIO_CHANNEL_LAYOUT_H_
+
+namespace webrtc {
+
+// This file is derived from Chromium's base/channel_layout.h.
+
+// Enumerates the various representations of the ordering of audio channels.
+// Logged to UMA, so never reuse a value, always add new/greater ones!
+enum ChannelLayout {
+  CHANNEL_LAYOUT_NONE = 0,
+  CHANNEL_LAYOUT_UNSUPPORTED = 1,
+
+  // Front C
+  CHANNEL_LAYOUT_MONO = 2,
+
+  // Front L, Front R
+  CHANNEL_LAYOUT_STEREO = 3,
+
+  // Front L, Front R, Back C
+  CHANNEL_LAYOUT_2_1 = 4,
+
+  // Front L, Front R, Front C
+  CHANNEL_LAYOUT_SURROUND = 5,
+
+  // Front L, Front R, Front C, Back C
+  CHANNEL_LAYOUT_4_0 = 6,
+
+  // Front L, Front R, Side L, Side R
+  CHANNEL_LAYOUT_2_2 = 7,
+
+  // Front L, Front R, Back L, Back R
+  CHANNEL_LAYOUT_QUAD = 8,
+
+  // Front L, Front R, Front C, Side L, Side R
+  CHANNEL_LAYOUT_5_0 = 9,
+
+  // Front L, Front R, Front C, LFE, Side L, Side R
+  CHANNEL_LAYOUT_5_1 = 10,
+
+  // Front L, Front R, Front C, Back L, Back R
+  CHANNEL_LAYOUT_5_0_BACK = 11,
+
+  // Front L, Front R, Front C, LFE, Back L, Back R
+  CHANNEL_LAYOUT_5_1_BACK = 12,
+
+  // Front L, Front R, Front C, Side L, Side R, Back L, Back R
+  CHANNEL_LAYOUT_7_0 = 13,
+
+  // Front L, Front R, Front C, LFE, Side L, Side R, Back L, Back R
+  CHANNEL_LAYOUT_7_1 = 14,
+
+  // Front L, Front R, Front C, LFE, Side L, Side R, Front LofC, Front RofC
+  CHANNEL_LAYOUT_7_1_WIDE = 15,
+
+  // Stereo L, Stereo R
+  CHANNEL_LAYOUT_STEREO_DOWNMIX = 16,
+
+  // Stereo L, Stereo R, LFE
+  CHANNEL_LAYOUT_2POINT1 = 17,
+
+  // Stereo L, Stereo R, Front C, LFE
+  CHANNEL_LAYOUT_3_1 = 18,
+
+  // Stereo L, Stereo R, Front C, Rear C, LFE
+  CHANNEL_LAYOUT_4_1 = 19,
+
+  // Stereo L, Stereo R, Front C, Side L, Side R, Back C
+  CHANNEL_LAYOUT_6_0 = 20,
+
+  // Stereo L, Stereo R, Side L, Side R, Front LofC, Front RofC
+  CHANNEL_LAYOUT_6_0_FRONT = 21,
+
+  // Stereo L, Stereo R, Front C, Rear L, Rear R, Rear C
+  CHANNEL_LAYOUT_HEXAGONAL = 22,
+
+  // Stereo L, Stereo R, Front C, LFE, Side L, Side R, Rear Center
+  CHANNEL_LAYOUT_6_1 = 23,
+
+  // Stereo L, Stereo R, Front C, LFE, Back L, Back R, Rear Center
+  CHANNEL_LAYOUT_6_1_BACK = 24,
+
+  // Stereo L, Stereo R, Side L, Side R, Front LofC, Front RofC, LFE
+  CHANNEL_LAYOUT_6_1_FRONT = 25,
+
+  // Front L, Front R, Front C, Side L, Side R, Front LofC, Front RofC
+  CHANNEL_LAYOUT_7_0_FRONT = 26,
+
+  // Front L, Front R, Front C, LFE, Back L, Back R, Front LofC, Front RofC
+  CHANNEL_LAYOUT_7_1_WIDE_BACK = 27,
+
+  // Front L, Front R, Front C, Side L, Side R, Rear L, Back R, Back C.
+  CHANNEL_LAYOUT_OCTAGONAL = 28,
+
+  // Channels are not explicitly mapped to speakers.
+  CHANNEL_LAYOUT_DISCRETE = 29,
+
+  // Front L, Front R, Front C. Front C contains the keyboard mic audio. This
+  // layout is only intended for input for WebRTC. The Front C channel
+  // is stripped away in the WebRTC audio input pipeline and never seen outside
+  // of that.
+  CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC = 30,
+
+  // Front L, Front R, Side L, Side R, LFE
+  CHANNEL_LAYOUT_4_1_QUAD_SIDE = 31,
+
+  // Actual channel layout is specified in the bitstream and the actual channel
+  // count is unknown at Chromium media pipeline level (useful for audio
+  // pass-through mode).
+  CHANNEL_LAYOUT_BITSTREAM = 32,
+
+  // Max value, must always equal the largest entry ever logged.
+  CHANNEL_LAYOUT_MAX = CHANNEL_LAYOUT_BITSTREAM
+};
+
+// Note: Do not reorder or reassign these values; other code depends on their
+// ordering to operate correctly. E.g., CoreAudio channel layout computations.
+enum Channels {
+  LEFT = 0,
+  RIGHT,
+  CENTER,
+  LFE,
+  BACK_LEFT,
+  BACK_RIGHT,
+  LEFT_OF_CENTER,
+  RIGHT_OF_CENTER,
+  BACK_CENTER,
+  SIDE_LEFT,
+  SIDE_RIGHT,
+  CHANNELS_MAX =
+      SIDE_RIGHT,  // Must always equal the largest value ever logged.
+};
+
+// The maximum number of concurrently active channels for all possible layouts.
+// ChannelLayoutToChannelCount() will never return a value higher than this.
+constexpr int kMaxConcurrentChannels = 8;
+
+// Returns the expected channel position in an interleaved stream.  Values of -1
+// mean the channel at that index is not used for that layout.  Values range
+// from 0 to ChannelLayoutToChannelCount(layout) - 1.
+int ChannelOrder(ChannelLayout layout, Channels channel);
+
+// Returns the number of channels in a given ChannelLayout.
+int ChannelLayoutToChannelCount(ChannelLayout layout);
+
+// Given the number of channels, return the best layout,
+// or return CHANNEL_LAYOUT_UNSUPPORTED if there is no good match.
+ChannelLayout GuessChannelLayout(int channels);
+
+// Returns a string representation of the channel layout.
+const char* ChannelLayoutToString(ChannelLayout layout);
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CHANNEL_LAYOUT_H_

TMessagesProj/jni/voip/webrtc/api/audio/echo_canceller3_config.cc

@@ -0,0 +1,278 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "api/audio/echo_canceller3_config.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace {
+bool Limit(float* value, float min, float max) {
+  float clamped = rtc::SafeClamp(*value, min, max);
+  clamped = std::isfinite(clamped) ? clamped : min;
+  bool res = *value == clamped;
+  *value = clamped;
+  return res;
+}
+
+bool Limit(size_t* value, size_t min, size_t max) {
+  size_t clamped = rtc::SafeClamp(*value, min, max);
+  bool res = *value == clamped;
+  *value = clamped;
+  return res;
+}
+
+bool Limit(int* value, int min, int max) {
+  int clamped = rtc::SafeClamp(*value, min, max);
+  bool res = *value == clamped;
+  *value = clamped;
+  return res;
+}
+
+bool FloorLimit(size_t* value, size_t min) {
+  size_t clamped = *value >= min ? *value : min;
+  bool res = *value == clamped;
+  *value = clamped;
+  return res;
+}
+
+}  // namespace
+
+EchoCanceller3Config::EchoCanceller3Config() = default;
+EchoCanceller3Config::EchoCanceller3Config(const EchoCanceller3Config& e) =
+    default;
+EchoCanceller3Config& EchoCanceller3Config::operator=(
+    const EchoCanceller3Config& e) = default;
+EchoCanceller3Config::Delay::Delay() = default;
+EchoCanceller3Config::Delay::Delay(const EchoCanceller3Config::Delay& e) =
+    default;
+EchoCanceller3Config::Delay& EchoCanceller3Config::Delay::operator=(
+    const Delay& e) = default;
+
+EchoCanceller3Config::EchoModel::EchoModel() = default;
+EchoCanceller3Config::EchoModel::EchoModel(
+    const EchoCanceller3Config::EchoModel& e) = default;
+EchoCanceller3Config::EchoModel& EchoCanceller3Config::EchoModel::operator=(
+    const EchoModel& e) = default;
+
+EchoCanceller3Config::Suppressor::Suppressor() = default;
+EchoCanceller3Config::Suppressor::Suppressor(
+    const EchoCanceller3Config::Suppressor& e) = default;
+EchoCanceller3Config::Suppressor& EchoCanceller3Config::Suppressor::operator=(
+    const Suppressor& e) = default;
+
+EchoCanceller3Config::Suppressor::MaskingThresholds::MaskingThresholds(
+    float enr_transparent,
+    float enr_suppress,
+    float emr_transparent)
+    : enr_transparent(enr_transparent),
+      enr_suppress(enr_suppress),
+      emr_transparent(emr_transparent) {}
+EchoCanceller3Config::Suppressor::MaskingThresholds::MaskingThresholds(
+    const EchoCanceller3Config::Suppressor::MaskingThresholds& e) = default;
+EchoCanceller3Config::Suppressor::MaskingThresholds&
+EchoCanceller3Config::Suppressor::MaskingThresholds::operator=(
+    const MaskingThresholds& e) = default;
+
+EchoCanceller3Config::Suppressor::Tuning::Tuning(MaskingThresholds mask_lf,
+                                                 MaskingThresholds mask_hf,
+                                                 float max_inc_factor,
+                                                 float max_dec_factor_lf)
+    : mask_lf(mask_lf),
+      mask_hf(mask_hf),
+      max_inc_factor(max_inc_factor),
+      max_dec_factor_lf(max_dec_factor_lf) {}
+EchoCanceller3Config::Suppressor::Tuning::Tuning(
+    const EchoCanceller3Config::Suppressor::Tuning& e) = default;
+EchoCanceller3Config::Suppressor::Tuning&
+EchoCanceller3Config::Suppressor::Tuning::operator=(const Tuning& e) = default;
+
+bool EchoCanceller3Config::Validate(EchoCanceller3Config* config) {
+  RTC_DCHECK(config);
+  EchoCanceller3Config* c = config;
+  bool res = true;
+
+  if (c->delay.down_sampling_factor != 4 &&
+      c->delay.down_sampling_factor != 8) {
+    c->delay.down_sampling_factor = 4;
+    res = false;
+  }
+
+  res = res & Limit(&c->delay.default_delay, 0, 5000);
+  res = res & Limit(&c->delay.num_filters, 0, 5000);
+  res = res & Limit(&c->delay.delay_headroom_samples, 0, 5000);
+  res = res & Limit(&c->delay.hysteresis_limit_blocks, 0, 5000);
+  res = res & Limit(&c->delay.fixed_capture_delay_samples, 0, 5000);
+  res = res & Limit(&c->delay.delay_estimate_smoothing, 0.f, 1.f);
+  res = res & Limit(&c->delay.delay_candidate_detection_threshold, 0.f, 1.f);
+  res = res & Limit(&c->delay.delay_selection_thresholds.initial, 1, 250);
+  res = res & Limit(&c->delay.delay_selection_thresholds.converged, 1, 250);
+
+  res = res & FloorLimit(&c->filter.refined.length_blocks, 1);
+  res = res & Limit(&c->filter.refined.leakage_converged, 0.f, 1000.f);
+  res = res & Limit(&c->filter.refined.leakage_diverged, 0.f, 1000.f);
+  res = res & Limit(&c->filter.refined.error_floor, 0.f, 1000.f);
+  res = res & Limit(&c->filter.refined.error_ceil, 0.f, 100000000.f);
+  res = res & Limit(&c->filter.refined.noise_gate, 0.f, 100000000.f);
+
+  res = res & FloorLimit(&c->filter.refined_initial.length_blocks, 1);
+  res = res & Limit(&c->filter.refined_initial.leakage_converged, 0.f, 1000.f);
+  res = res & Limit(&c->filter.refined_initial.leakage_diverged, 0.f, 1000.f);
+  res = res & Limit(&c->filter.refined_initial.error_floor, 0.f, 1000.f);
+  res = res & Limit(&c->filter.refined_initial.error_ceil, 0.f, 100000000.f);
+  res = res & Limit(&c->filter.refined_initial.noise_gate, 0.f, 100000000.f);
+
+  if (c->filter.refined.length_blocks <
+      c->filter.refined_initial.length_blocks) {
+    c->filter.refined_initial.length_blocks = c->filter.refined.length_blocks;
+    res = false;
+  }
+
+  res = res & FloorLimit(&c->filter.coarse.length_blocks, 1);
+  res = res & Limit(&c->filter.coarse.rate, 0.f, 1.f);
+  res = res & Limit(&c->filter.coarse.noise_gate, 0.f, 100000000.f);
+
+  res = res & FloorLimit(&c->filter.coarse_initial.length_blocks, 1);
+  res = res & Limit(&c->filter.coarse_initial.rate, 0.f, 1.f);
+  res = res & Limit(&c->filter.coarse_initial.noise_gate, 0.f, 100000000.f);
+
+  if (c->filter.coarse.length_blocks < c->filter.coarse_initial.length_blocks) {
+    c->filter.coarse_initial.length_blocks = c->filter.coarse.length_blocks;
+    res = false;
+  }
+
+  res = res & Limit(&c->filter.config_change_duration_blocks, 0, 100000);
+  res = res & Limit(&c->filter.initial_state_seconds, 0.f, 100.f);
+  res = res & Limit(&c->filter.coarse_reset_hangover_blocks, 0, 250000);
+
+  res = res & Limit(&c->erle.min, 1.f, 100000.f);
+  res = res & Limit(&c->erle.max_l, 1.f, 100000.f);
+  res = res & Limit(&c->erle.max_h, 1.f, 100000.f);
+  if (c->erle.min > c->erle.max_l || c->erle.min > c->erle.max_h) {
+    c->erle.min = std::min(c->erle.max_l, c->erle.max_h);
+    res = false;
+  }
+  res = res & Limit(&c->erle.num_sections, 1, c->filter.refined.length_blocks);
+
+  res = res & Limit(&c->ep_strength.default_gain, 0.f, 1000000.f);
+  res = res & Limit(&c->ep_strength.default_len, -1.f, 1.f);
+  res = res & Limit(&c->ep_strength.nearend_len, -1.0f, 1.0f);
+
+  res =
+      res & Limit(&c->echo_audibility.low_render_limit, 0.f, 32768.f * 32768.f);
+  res = res &
+        Limit(&c->echo_audibility.normal_render_limit, 0.f, 32768.f * 32768.f);
+  res = res & Limit(&c->echo_audibility.floor_power, 0.f, 32768.f * 32768.f);
+  res = res & Limit(&c->echo_audibility.audibility_threshold_lf, 0.f,
+                    32768.f * 32768.f);
+  res = res & Limit(&c->echo_audibility.audibility_threshold_mf, 0.f,
+                    32768.f * 32768.f);
+  res = res & Limit(&c->echo_audibility.audibility_threshold_hf, 0.f,
+                    32768.f * 32768.f);
+
+  res = res &
+        Limit(&c->render_levels.active_render_limit, 0.f, 32768.f * 32768.f);
+  res = res & Limit(&c->render_levels.poor_excitation_render_limit, 0.f,
+                    32768.f * 32768.f);
+  res = res & Limit(&c->render_levels.poor_excitation_render_limit_ds8, 0.f,
+                    32768.f * 32768.f);
+
+  res = res & Limit(&c->echo_model.noise_floor_hold, 0, 1000);
+  res = res & Limit(&c->echo_model.min_noise_floor_power, 0, 2000000.f);
+  res = res & Limit(&c->echo_model.stationary_gate_slope, 0, 1000000.f);
+  res = res & Limit(&c->echo_model.noise_gate_power, 0, 1000000.f);
+  res = res & Limit(&c->echo_model.noise_gate_slope, 0, 1000000.f);
+  res = res & Limit(&c->echo_model.render_pre_window_size, 0, 100);
+  res = res & Limit(&c->echo_model.render_post_window_size, 0, 100);
+
+  res = res & Limit(&c->comfort_noise.noise_floor_dbfs, -200.f, 0.f);
+
+  res = res & Limit(&c->suppressor.nearend_average_blocks, 1, 5000);
+
+  res = res &
+        Limit(&c->suppressor.normal_tuning.mask_lf.enr_transparent, 0.f, 100.f);
+  res = res &
+        Limit(&c->suppressor.normal_tuning.mask_lf.enr_suppress, 0.f, 100.f);
+  res = res &
+        Limit(&c->suppressor.normal_tuning.mask_lf.emr_transparent, 0.f, 100.f);
+  res = res &
+        Limit(&c->suppressor.normal_tuning.mask_hf.enr_transparent, 0.f, 100.f);
+  res = res &
+        Limit(&c->suppressor.normal_tuning.mask_hf.enr_suppress, 0.f, 100.f);
+  res = res &
+        Limit(&c->suppressor.normal_tuning.mask_hf.emr_transparent, 0.f, 100.f);
+  res = res & Limit(&c->suppressor.normal_tuning.max_inc_factor, 0.f, 100.f);
+  res = res & Limit(&c->suppressor.normal_tuning.max_dec_factor_lf, 0.f, 100.f);
+
+  res = res & Limit(&c->suppressor.nearend_tuning.mask_lf.enr_transparent, 0.f,
+                    100.f);
+  res = res &
+        Limit(&c->suppressor.nearend_tuning.mask_lf.enr_suppress, 0.f, 100.f);
+  res = res & Limit(&c->suppressor.nearend_tuning.mask_lf.emr_transparent, 0.f,
+                    100.f);
+  res = res & Limit(&c->suppressor.nearend_tuning.mask_hf.enr_transparent, 0.f,
+                    100.f);
+  res = res &
+        Limit(&c->suppressor.nearend_tuning.mask_hf.enr_suppress, 0.f, 100.f);
+  res = res & Limit(&c->suppressor.nearend_tuning.mask_hf.emr_transparent, 0.f,
+                    100.f);
+  res = res & Limit(&c->suppressor.nearend_tuning.max_inc_factor, 0.f, 100.f);
+  res =
+      res & Limit(&c->suppressor.nearend_tuning.max_dec_factor_lf, 0.f, 100.f);
+
+  res = res & Limit(&c->suppressor.last_permanent_lf_smoothing_band, 0, 64);
+  res = res & Limit(&c->suppressor.last_lf_smoothing_band, 0, 64);
+  res = res & Limit(&c->suppressor.last_lf_band, 0, 63);
+  res = res &
+        Limit(&c->suppressor.first_hf_band, c->suppressor.last_lf_band + 1, 64);
+
+  res = res & Limit(&c->suppressor.dominant_nearend_detection.enr_threshold,
+                    0.f, 1000000.f);
+  res = res & Limit(&c->suppressor.dominant_nearend_detection.snr_threshold,
+                    0.f, 1000000.f);
+  res = res & Limit(&c->suppressor.dominant_nearend_detection.hold_duration, 0,
+                    10000);
+  res = res & Limit(&c->suppressor.dominant_nearend_detection.trigger_threshold,
+                    0, 10000);
+
+  res = res &
+        Limit(&c->suppressor.subband_nearend_detection.nearend_average_blocks,
+              1, 1024);
+  res =
+      res & Limit(&c->suppressor.subband_nearend_detection.subband1.low, 0, 65);
+  res = res & Limit(&c->suppressor.subband_nearend_detection.subband1.high,
+                    c->suppressor.subband_nearend_detection.subband1.low, 65);
+  res =
+      res & Limit(&c->suppressor.subband_nearend_detection.subband2.low, 0, 65);
+  res = res & Limit(&c->suppressor.subband_nearend_detection.subband2.high,
+                    c->suppressor.subband_nearend_detection.subband2.low, 65);
+  res = res & Limit(&c->suppressor.subband_nearend_detection.nearend_threshold,
+                    0.f, 1.e24f);
+  res = res & Limit(&c->suppressor.subband_nearend_detection.snr_threshold, 0.f,
+                    1.e24f);
+
+  res = res & Limit(&c->suppressor.high_bands_suppression.enr_threshold, 0.f,
+                    1000000.f);
+  res = res & Limit(&c->suppressor.high_bands_suppression.max_gain_during_echo,
+                    0.f, 1.f);
+  res = res & Limit(&c->suppressor.high_bands_suppression
+                         .anti_howling_activation_threshold,
+                    0.f, 32768.f * 32768.f);
+  res = res & Limit(&c->suppressor.high_bands_suppression.anti_howling_gain,
+                    0.f, 1.f);
+
+  res = res & Limit(&c->suppressor.floor_first_increase, 0.f, 1000000.f);
+
+  return res;
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio/echo_canceller3_config.h

@@ -0,0 +1,250 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_ECHO_CANCELLER3_CONFIG_H_
+#define API_AUDIO_ECHO_CANCELLER3_CONFIG_H_
+
+#include <stddef.h>  // size_t
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Configuration struct for EchoCanceller3
+struct RTC_EXPORT EchoCanceller3Config {
+  // Checks and updates the config parameters to lie within (mostly) reasonable
+  // ranges. Returns true if and only of the config did not need to be changed.
+  static bool Validate(EchoCanceller3Config* config);
+
+  EchoCanceller3Config();
+  EchoCanceller3Config(const EchoCanceller3Config& e);
+  EchoCanceller3Config& operator=(const EchoCanceller3Config& other);
+
+  struct Buffering {
+    size_t excess_render_detection_interval_blocks = 250;
+    size_t max_allowed_excess_render_blocks = 8;
+  } buffering;
+
+  struct Delay {
+    Delay();
+    Delay(const Delay& e);
+    Delay& operator=(const Delay& e);
+    size_t default_delay = 5;
+    size_t down_sampling_factor = 4;
+    size_t num_filters = 5;
+    size_t delay_headroom_samples = 32;
+    size_t hysteresis_limit_blocks = 1;
+    size_t fixed_capture_delay_samples = 0;
+    float delay_estimate_smoothing = 0.7f;
+    float delay_estimate_smoothing_delay_found = 0.7f;
+    float delay_candidate_detection_threshold = 0.2f;
+    struct DelaySelectionThresholds {
+      int initial;
+      int converged;
+    } delay_selection_thresholds = {5, 20};
+    bool use_external_delay_estimator = false;
+    bool log_warning_on_delay_changes = false;
+    struct AlignmentMixing {
+      bool downmix;
+      bool adaptive_selection;
+      float activity_power_threshold;
+      bool prefer_first_two_channels;
+    };
+    AlignmentMixing render_alignment_mixing = {false, true, 10000.f, true};
+    AlignmentMixing capture_alignment_mixing = {false, true, 10000.f, false};
+    bool detect_pre_echo = true;
+  } delay;
+
+  struct Filter {
+    struct RefinedConfiguration {
+      size_t length_blocks;
+      float leakage_converged;
+      float leakage_diverged;
+      float error_floor;
+      float error_ceil;
+      float noise_gate;
+    };
+
+    struct CoarseConfiguration {
+      size_t length_blocks;
+      float rate;
+      float noise_gate;
+    };
+
+    RefinedConfiguration refined = {13,     0.00005f, 0.05f,
+                                    0.001f, 2.f,      20075344.f};
+    CoarseConfiguration coarse = {13, 0.7f, 20075344.f};
+
+    RefinedConfiguration refined_initial = {12,     0.005f, 0.5f,
+                                            0.001f, 2.f,    20075344.f};
+    CoarseConfiguration coarse_initial = {12, 0.9f, 20075344.f};
+
+    size_t config_change_duration_blocks = 250;
+    float initial_state_seconds = 2.5f;
+    int coarse_reset_hangover_blocks = 25;
+    bool conservative_initial_phase = false;
+    bool enable_coarse_filter_output_usage = true;
+    bool use_linear_filter = true;
+    bool high_pass_filter_echo_reference = false;
+    bool export_linear_aec_output = false;
+  } filter;
+
+  struct Erle {
+    float min = 1.f;
+    float max_l = 4.f;
+    float max_h = 1.5f;
+    bool onset_detection = true;
+    size_t num_sections = 1;
+    bool clamp_quality_estimate_to_zero = true;
+    bool clamp_quality_estimate_to_one = true;
+  } erle;
+
+  struct EpStrength {
+    float default_gain = 1.f;
+    float default_len = 0.83f;
+    float nearend_len = 0.83f;
+    bool echo_can_saturate = true;
+    bool bounded_erl = false;
+    bool erle_onset_compensation_in_dominant_nearend = false;
+    bool use_conservative_tail_frequency_response = true;
+  } ep_strength;
+
+  struct EchoAudibility {
+    float low_render_limit = 4 * 64.f;
+    float normal_render_limit = 64.f;
+    float floor_power = 2 * 64.f;
+    float audibility_threshold_lf = 10;
+    float audibility_threshold_mf = 10;
+    float audibility_threshold_hf = 10;
+    bool use_stationarity_properties = false;
+    bool use_stationarity_properties_at_init = false;
+  } echo_audibility;
+
+  struct RenderLevels {
+    float active_render_limit = 100.f;
+    float poor_excitation_render_limit = 150.f;
+    float poor_excitation_render_limit_ds8 = 20.f;
+    float render_power_gain_db = 0.f;
+  } render_levels;
+
+  struct EchoRemovalControl {
+    bool has_clock_drift = false;
+    bool linear_and_stable_echo_path = false;
+  } echo_removal_control;
+
+  struct EchoModel {
+    EchoModel();
+    EchoModel(const EchoModel& e);
+    EchoModel& operator=(const EchoModel& e);
+    size_t noise_floor_hold = 50;
+    float min_noise_floor_power = 1638400.f;
+    float stationary_gate_slope = 10.f;
+    float noise_gate_power = 27509.42f;
+    float noise_gate_slope = 0.3f;
+    size_t render_pre_window_size = 1;
+    size_t render_post_window_size = 1;
+    bool model_reverb_in_nonlinear_mode = true;
+  } echo_model;
+
+  struct ComfortNoise {
+    float noise_floor_dbfs = -96.03406f;
+  } comfort_noise;
+
+  struct Suppressor {
+    Suppressor();
+    Suppressor(const Suppressor& e);
+    Suppressor& operator=(const Suppressor& e);
+
+    size_t nearend_average_blocks = 4;
+
+    struct MaskingThresholds {
+      MaskingThresholds(float enr_transparent,
+                        float enr_suppress,
+                        float emr_transparent);
+      MaskingThresholds(const MaskingThresholds& e);
+      MaskingThresholds& operator=(const MaskingThresholds& e);
+      float enr_transparent;
+      float enr_suppress;
+      float emr_transparent;
+    };
+
+    struct Tuning {
+      Tuning(MaskingThresholds mask_lf,
+             MaskingThresholds mask_hf,
+             float max_inc_factor,
+             float max_dec_factor_lf);
+      Tuning(const Tuning& e);
+      Tuning& operator=(const Tuning& e);
+      MaskingThresholds mask_lf;
+      MaskingThresholds mask_hf;
+      float max_inc_factor;
+      float max_dec_factor_lf;
+    };
+
+    Tuning normal_tuning = Tuning(MaskingThresholds(.3f, .4f, .3f),
+                                  MaskingThresholds(.07f, .1f, .3f),
+                                  2.0f,
+                                  0.25f);
+    Tuning nearend_tuning = Tuning(MaskingThresholds(1.09f, 1.1f, .3f),
+                                   MaskingThresholds(.1f, .3f, .3f),
+                                   2.0f,
+                                   0.25f);
+
+    bool lf_smoothing_during_initial_phase = true;
+    int last_permanent_lf_smoothing_band = 0;
+    int last_lf_smoothing_band = 5;
+    int last_lf_band = 5;
+    int first_hf_band = 8;
+
+    struct DominantNearendDetection {
+      float enr_threshold = .25f;
+      float enr_exit_threshold = 10.f;
+      float snr_threshold = 30.f;
+      int hold_duration = 50;
+      int trigger_threshold = 12;
+      bool use_during_initial_phase = true;
+      bool use_unbounded_echo_spectrum = true;
+    } dominant_nearend_detection;
+
+    struct SubbandNearendDetection {
+      size_t nearend_average_blocks = 1;
+      struct SubbandRegion {
+        size_t low;
+        size_t high;
+      };
+      SubbandRegion subband1 = {1, 1};
+      SubbandRegion subband2 = {1, 1};
+      float nearend_threshold = 1.f;
+      float snr_threshold = 1.f;
+    } subband_nearend_detection;
+
+    bool use_subband_nearend_detection = false;
+
+    struct HighBandsSuppression {
+      float enr_threshold = 1.f;
+      float max_gain_during_echo = 1.f;
+      float anti_howling_activation_threshold = 400.f;
+      float anti_howling_gain = 1.f;
+    } high_bands_suppression;
+
+    float floor_first_increase = 0.00001f;
+    bool conservative_hf_suppression = false;
+  } suppressor;
+
+  struct MultiChannel {
+    bool detect_stereo_content = true;
+    float stereo_detection_threshold = 0.0f;
+    int stereo_detection_timeout_threshold_seconds = 300;
+    float stereo_detection_hysteresis_seconds = 2.0f;
+  } multi_channel;
+};
+}  // namespace webrtc
+
+#endif  // API_AUDIO_ECHO_CANCELLER3_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio/echo_canceller3_factory.cc

@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "api/audio/echo_canceller3_factory.h"
+
+#include <memory>
+
+#include "modules/audio_processing/aec3/echo_canceller3.h"
+
+namespace webrtc {
+
+EchoCanceller3Factory::EchoCanceller3Factory() {}
+
+EchoCanceller3Factory::EchoCanceller3Factory(const EchoCanceller3Config& config)
+    : config_(config) {}
+
+std::unique_ptr<EchoControl> EchoCanceller3Factory::Create(
+    int sample_rate_hz,
+    int num_render_channels,
+    int num_capture_channels) {
+  return std::make_unique<EchoCanceller3>(
+      config_, /*multichannel_config=*/absl::nullopt, sample_rate_hz,
+      num_render_channels, num_capture_channels);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio/echo_canceller3_factory.h

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
+#define API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
+
+#include <memory>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_control.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RTC_EXPORT EchoCanceller3Factory : public EchoControlFactory {
+ public:
+  // Factory producing EchoCanceller3 instances with the default configuration.
+  EchoCanceller3Factory();
+
+  // Factory producing EchoCanceller3 instances with the specified
+  // configuration.
+  explicit EchoCanceller3Factory(const EchoCanceller3Config& config);
+
+  // Creates an EchoCanceller3 with a specified channel count and sampling rate.
+  std::unique_ptr<EchoControl> Create(int sample_rate_hz,
+                                      int num_render_channels,
+                                      int num_capture_channels) override;
+
+ private:
+  const EchoCanceller3Config config_;
+};
+}  // namespace webrtc
+
+#endif  // API_AUDIO_ECHO_CANCELLER3_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/audio/echo_control.h

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_ECHO_CONTROL_H_
+#define API_AUDIO_ECHO_CONTROL_H_
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioBuffer;
+
+// Interface for an acoustic echo cancellation (AEC) submodule.
+class EchoControl {
+ public:
+  // Analysis (not changing) of the render signal.
+  virtual void AnalyzeRender(AudioBuffer* render) = 0;
+
+  // Analysis (not changing) of the capture signal.
+  virtual void AnalyzeCapture(AudioBuffer* capture) = 0;
+
+  // Processes the capture signal in order to remove the echo.
+  virtual void ProcessCapture(AudioBuffer* capture, bool level_change) = 0;
+
+  // As above, but also returns the linear filter output.
+  virtual void ProcessCapture(AudioBuffer* capture,
+                              AudioBuffer* linear_output,
+                              bool level_change) = 0;
+
+  struct Metrics {
+    double echo_return_loss;
+    double echo_return_loss_enhancement;
+    int delay_ms;
+  };
+
+  // Collect current metrics from the echo controller.
+  virtual Metrics GetMetrics() const = 0;
+
+  // Provides an optional external estimate of the audio buffer delay.
+  virtual void SetAudioBufferDelay(int delay_ms) = 0;
+
+  // Specifies whether the capture output will be used. The purpose of this is
+  // to allow the echo controller to deactivate some of the processing when the
+  // resulting output is anyway not used, for instance when the endpoint is
+  // muted.
+  // TODO(b/177830919): Make pure virtual.
+  virtual void SetCaptureOutputUsage(bool capture_output_used) {}
+
+  // Returns wheter the signal is altered.
+  virtual bool ActiveProcessing() const = 0;
+
+  virtual ~EchoControl() {}
+};
+
+// Interface for a factory that creates EchoControllers.
+class EchoControlFactory {
+ public:
+  virtual std::unique_ptr<EchoControl> Create(int sample_rate_hz,
+                                              int num_render_channels,
+                                              int num_capture_channels) = 0;
+
+  virtual ~EchoControlFactory() = default;
+};
+}  // namespace webrtc
+
+#endif  // API_AUDIO_ECHO_CONTROL_H_

TMessagesProj/jni/voip/webrtc/api/audio/echo_detector_creator.cc

@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "api/audio/echo_detector_creator.h"
+
+#include "api/make_ref_counted.h"
+#include "modules/audio_processing/residual_echo_detector.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<EchoDetector> CreateEchoDetector() {
+  return rtc::make_ref_counted<ResidualEchoDetector>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio/echo_detector_creator.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_ECHO_DETECTOR_CREATOR_H_
+#define API_AUDIO_ECHO_DETECTOR_CREATOR_H_
+
+#include "api/scoped_refptr.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+// Returns an instance of the WebRTC implementation of a residual echo detector.
+// It can be provided to the webrtc::AudioProcessingBuilder to obtain the
+// usual residual echo metrics.
+rtc::scoped_refptr<EchoDetector> CreateEchoDetector();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_ECHO_DETECTOR_CREATOR_H_

TMessagesProj/jni/voip/webrtc/api/audio/test/BUILD.gn

@@ -0,0 +1,29 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+if (rtc_include_tests) {
+  rtc_library("audio_api_unittests") {
+    testonly = true
+    sources = [
+      "audio_frame_unittest.cc",
+      "echo_canceller3_config_unittest.cc",
+    ]
+    deps = [
+      "..:aec3_config",
+      "..:audio_frame_api",
+      "../../../modules/audio_processing:aec3_config_json",
+      "../../../test:test_support",
+    ]
+  }
+}

TMessagesProj/jni/voip/webrtc/api/audio/test/audio_frame_unittest.cc

@@ -0,0 +1,136 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio/audio_frame.h"
+
+#include <stdint.h>
+#include <string.h>  // memcmp
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
+  const int16_t* frame_data = frame.data();
+  for (size_t i = 0; i < frame.max_16bit_samples(); i++) {
+    if (frame_data[i] != sample) {
+      return false;
+    }
+  }
+  return true;
+}
+
+constexpr uint32_t kTimestamp = 27;
+constexpr int kSampleRateHz = 16000;
+constexpr size_t kNumChannelsMono = 1;
+constexpr size_t kNumChannelsStereo = 2;
+constexpr size_t kNumChannels5_1 = 6;
+constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
+
+}  // namespace
+
+TEST(AudioFrameTest, FrameStartsMuted) {
+  AudioFrame frame;
+  EXPECT_TRUE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
+  AudioFrame frame;
+  frame.mutable_data();
+  EXPECT_FALSE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
+  AudioFrame frame;
+  int16_t* frame_data = frame.mutable_data();
+  for (size_t i = 0; i < frame.max_16bit_samples(); i++) {
+    frame_data[i] = 17;
+  }
+  ASSERT_TRUE(AllSamplesAre(17, frame));
+  frame.Mute();
+  EXPECT_TRUE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, UpdateFrameMono) {
+  AudioFrame frame;
+  int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17};
+  frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
+                    AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannelsMono);
+
+  EXPECT_EQ(kTimestamp, frame.timestamp_);
+  EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
+  EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz());
+  EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
+  EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
+  EXPECT_EQ(kNumChannelsMono, frame.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame.channel_layout());
+
+  EXPECT_FALSE(frame.muted());
+  EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
+
+  frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
+                    kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+                    kNumChannelsMono);
+  EXPECT_TRUE(frame.muted());
+  EXPECT_TRUE(AllSamplesAre(0, frame));
+}
+
+TEST(AudioFrameTest, UpdateFrameMultiChannel) {
+  AudioFrame frame;
+  frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
+                    kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+                    kNumChannelsStereo);
+  EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
+  EXPECT_EQ(kNumChannelsStereo, frame.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame.channel_layout());
+
+  frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
+                    kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+                    kNumChannels5_1);
+  EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel());
+  EXPECT_EQ(kNumChannels5_1, frame.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_5_1, frame.channel_layout());
+}
+
+TEST(AudioFrameTest, CopyFrom) {
+  AudioFrame frame1;
+  AudioFrame frame2;
+
+  int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17};
+  frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
+                     AudioFrame::kPLC, AudioFrame::kVadActive,
+                     kNumChannelsMono);
+  frame1.CopyFrom(frame2);
+
+  EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
+  EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
+  EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
+  EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
+  EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
+  EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
+
+  EXPECT_EQ(frame2.muted(), frame1.muted());
+  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
+
+  frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
+                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
+                     kNumChannelsMono);
+  frame1.CopyFrom(frame2);
+
+  EXPECT_EQ(frame2.muted(), frame1.muted());
+  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio/test/echo_canceller3_config_unittest.cc

@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio/echo_canceller3_config.h"
+
+#include "modules/audio_processing/test/echo_canceller3_config_json.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(EchoCanceller3Config, ValidConfigIsNotModified) {
+  EchoCanceller3Config config;
+  EXPECT_TRUE(EchoCanceller3Config::Validate(&config));
+  EchoCanceller3Config default_config;
+  EXPECT_EQ(Aec3ConfigToJsonString(config),
+            Aec3ConfigToJsonString(default_config));
+}
+
+TEST(EchoCanceller3Config, InvalidConfigIsCorrected) {
+  // Change a parameter and validate.
+  EchoCanceller3Config config;
+  config.echo_model.min_noise_floor_power = -1600000.f;
+  EXPECT_FALSE(EchoCanceller3Config::Validate(&config));
+  EXPECT_GE(config.echo_model.min_noise_floor_power, 0.f);
+  // Verify remaining parameters are unchanged.
+  EchoCanceller3Config default_config;
+  config.echo_model.min_noise_floor_power =
+      default_config.echo_model.min_noise_floor_power;
+  EXPECT_EQ(Aec3ConfigToJsonString(config),
+            Aec3ConfigToJsonString(default_config));
+}
+
+TEST(EchoCanceller3Config, ValidatedConfigsAreValid) {
+  EchoCanceller3Config config;
+  config.delay.down_sampling_factor = 983;
+  EXPECT_FALSE(EchoCanceller3Config::Validate(&config));
+  EXPECT_TRUE(EchoCanceller3Config::Validate(&config));
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/BUILD.gn

@@ -0,0 +1,147 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_codecs_api") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_codec_pair_id.cc",
+    "audio_codec_pair_id.h",
+    "audio_decoder.cc",
+    "audio_decoder.h",
+    "audio_decoder_factory.h",
+    "audio_decoder_factory_template.h",
+    "audio_encoder.cc",
+    "audio_encoder.h",
+    "audio_encoder_factory.h",
+    "audio_encoder_factory_template.h",
+    "audio_format.cc",
+    "audio_format.h",
+  ]
+  deps = [
+    "..:array_view",
+    "..:bitrate_allocation",
+    "..:make_ref_counted",
+    "..:ref_count",
+    "..:scoped_refptr",
+    "../../api:field_trials_view",
+    "../../api:rtp_parameters",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:refcount",
+    "../../rtc_base:sanitizer",
+    "../../rtc_base/system:rtc_export",
+    "../units:data_rate",
+    "../units:time_delta",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("builtin_audio_decoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "builtin_audio_decoder_factory.cc",
+    "builtin_audio_decoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "L16:audio_decoder_L16",
+    "g711:audio_decoder_g711",
+    "g722:audio_decoder_g722",
+  ]
+  defines = []
+  if (rtc_include_ilbc) {
+    deps += [ "ilbc:audio_decoder_ilbc" ]
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
+  }
+  if (rtc_include_opus) {
+    deps += [
+      "opus:audio_decoder_multiopus",
+      "opus:audio_decoder_opus",
+    ]
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
+  }
+}
+
+rtc_library("builtin_audio_encoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "builtin_audio_encoder_factory.cc",
+    "builtin_audio_encoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "L16:audio_encoder_L16",
+    "g711:audio_encoder_g711",
+    "g722:audio_encoder_g722",
+  ]
+  defines = []
+  if (rtc_include_ilbc) {
+    deps += [ "ilbc:audio_encoder_ilbc" ]
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
+  }
+  if (rtc_include_opus) {
+    deps += [
+      "opus:audio_encoder_multiopus",
+      "opus:audio_encoder_opus",
+    ]
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
+  }
+}
+
+rtc_library("opus_audio_decoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "opus_audio_decoder_factory.cc",
+    "opus_audio_decoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "opus:audio_decoder_multiopus",
+    "opus:audio_decoder_opus",
+  ]
+}
+
+rtc_library("opus_audio_encoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "opus_audio_encoder_factory.cc",
+    "opus_audio_encoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "opus:audio_encoder_multiopus",
+    "opus:audio_encoder_opus",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/BUILD.gn

@@ -0,0 +1,55 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_L16") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_L16.cc",
+    "audio_encoder_L16.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:pcm16b",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_L16") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_L16.cc",
+    "audio_decoder_L16.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:pcm16b",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc

@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/L16/audio_decoder_L16.h"
+
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
+    const SdpAudioFormat& format) {
+  Config config;
+  config.sample_rate_hz = format.clockrate_hz;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
+    return config;
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderL16::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  Pcm16BAppendSupportedCodecSpecs(specs);
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
+    const Config& config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderPcm16B>(config.sample_rate_hz,
+                                              config.num_channels);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/audio_decoder_L16.h

@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
+#define API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// L16 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderL16 {
+  struct Config {
+    bool IsOk() const {
+      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
+             (num_channels >= 1 &&
+              num_channels <= AudioDecoder::kMaxNumberOfChannels);
+    }
+    int sample_rate_hz = 8000;
+    int num_channels = 1;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc

@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  Config config;
+  config.sample_rate_hz = format.clockrate_hz;
+  config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+    }
+  }
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
+    return config;
+  }
+  return absl::nullopt;
+}
+
+void AudioEncoderL16::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  Pcm16BAppendSupportedCodecSpecs(specs);
+}
+
+AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
+    const AudioEncoderL16::Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {config.sample_rate_hz,
+          rtc::dchecked_cast<size_t>(config.num_channels),
+          config.sample_rate_hz * config.num_channels * 16};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
+    const AudioEncoderL16::Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  AudioEncoderPcm16B::Config c;
+  c.sample_rate_hz = config.sample_rate_hz;
+  c.num_channels = config.num_channels;
+  c.frame_size_ms = config.frame_size_ms;
+  c.payload_type = payload_type;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderPcm16B>(c);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/L16/audio_encoder_L16.h

@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
+#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// L16 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderL16 {
+  struct Config {
+    bool IsOk() const {
+      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels &&
+             frame_size_ms > 0 && frame_size_ms <= 120 &&
+             frame_size_ms % 10 == 0;
+    }
+    int sample_rate_hz = 8000;
+    int num_channels = 1;
+    int frame_size_ms = 10;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/OWNERS

@@ -0,0 +1,3 @@
+alessiob@webrtc.org
+henrik.lundin@webrtc.org
+jakobi@webrtc.org

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_codec_pair_id.cc

@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/audio_codec_pair_id.h"
+
+#include <atomic>
+#include <cstdint>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Returns a new value that it has never returned before. You may call it at
+// most 2^63 times in the lifetime of the program. Note: The returned values
+// may be easily predictable.
+uint64_t GetNextId() {
+  static std::atomic<uint64_t> next_id(0);
+
+  // Atomically increment `next_id`, and return the previous value. Relaxed
+  // memory order is sufficient, since all we care about is that different
+  // callers return different values.
+  const uint64_t new_id = next_id.fetch_add(1, std::memory_order_relaxed);
+
+  // This check isn't atomic with the increment, so if we start 2^63 + 1
+  // invocations of GetNextId() in parallel, the last one to do the atomic
+  // increment could return the ID 0 before any of the others had time to
+  // trigger this DCHECK. We blithely assume that this won't happen.
+  RTC_DCHECK_LT(new_id, uint64_t{1} << 63) << "Used up all ID values";
+
+  return new_id;
+}
+
+// Make an integer ID more unpredictable. This is a 1:1 mapping, so you can
+// feed it any value, but the idea is that you can feed it a sequence such as
+// 0, 1, 2, ... and get a new sequence that isn't as trivially predictable, so
+// that users won't rely on it being consecutive or increasing or anything like
+// that.
+constexpr uint64_t ObfuscateId(uint64_t id) {
+  // Any nonzero coefficient that's relatively prime to 2^64 (that is, any odd
+  // number) and any constant will give a 1:1 mapping. These high-entropy
+  // values will prevent the sequence from being trivially predictable.
+  //
+  // Both the multiplication and the addition going to overflow almost always,
+  // but that's fine---we *want* arithmetic mod 2^64.
+  return uint64_t{0x85fdb20e1294309a} + uint64_t{0xc516ef5c37462469} * id;
+}
+
+// The first ten values. Verified against the Python function
+//
+//   def f(n):
+//     return (0x85fdb20e1294309a + 0xc516ef5c37462469 * n) % 2**64
+//
+// Callers should obviously not depend on these exact values...
+//
+// (On Visual C++, we have to disable warning C4307 (integral constant
+// overflow), even though unsigned integers have perfectly well-defined
+// overflow behavior.)
+#ifdef _MSC_VER
+#pragma warning(push)
+#pragma warning(disable : 4307)
+#endif
+static_assert(ObfuscateId(0) == uint64_t{0x85fdb20e1294309a}, "");
+static_assert(ObfuscateId(1) == uint64_t{0x4b14a16a49da5503}, "");
+static_assert(ObfuscateId(2) == uint64_t{0x102b90c68120796c}, "");
+static_assert(ObfuscateId(3) == uint64_t{0xd5428022b8669dd5}, "");
+static_assert(ObfuscateId(4) == uint64_t{0x9a596f7eefacc23e}, "");
+static_assert(ObfuscateId(5) == uint64_t{0x5f705edb26f2e6a7}, "");
+static_assert(ObfuscateId(6) == uint64_t{0x24874e375e390b10}, "");
+static_assert(ObfuscateId(7) == uint64_t{0xe99e3d93957f2f79}, "");
+static_assert(ObfuscateId(8) == uint64_t{0xaeb52cefccc553e2}, "");
+static_assert(ObfuscateId(9) == uint64_t{0x73cc1c4c040b784b}, "");
+#ifdef _MSC_VER
+#pragma warning(pop)
+#endif
+
+}  // namespace
+
+AudioCodecPairId AudioCodecPairId::Create() {
+  return AudioCodecPairId(ObfuscateId(GetNextId()));
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_codec_pair_id.h

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
+#define API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
+
+#include <stdint.h>
+
+#include <utility>
+
+namespace webrtc {
+
+class AudioCodecPairId final {
+ public:
+  // Copyable, but not default constructible.
+  AudioCodecPairId() = delete;
+  AudioCodecPairId(const AudioCodecPairId&) = default;
+  AudioCodecPairId(AudioCodecPairId&&) = default;
+  AudioCodecPairId& operator=(const AudioCodecPairId&) = default;
+  AudioCodecPairId& operator=(AudioCodecPairId&&) = default;
+
+  friend void swap(AudioCodecPairId& a, AudioCodecPairId& b) {
+    using std::swap;
+    swap(a.id_, b.id_);
+  }
+
+  // Creates a new ID, unequal to any previously created ID.
+  static AudioCodecPairId Create();
+
+  // IDs can be tested for equality.
+  friend bool operator==(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ == b.id_;
+  }
+  friend bool operator!=(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ != b.id_;
+  }
+
+  // Comparisons. The ordering of ID values is completely arbitrary, but
+  // stable, so it's useful e.g. if you want to use IDs as keys in an ordered
+  // map.
+  friend bool operator<(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ < b.id_;
+  }
+  friend bool operator<=(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ <= b.id_;
+  }
+  friend bool operator>=(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ >= b.id_;
+  }
+  friend bool operator>(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ > b.id_;
+  }
+
+  // Returns a numeric representation of the ID. The numeric values are
+  // completely arbitrary, but stable, collision-free, and reasonably evenly
+  // distributed, so they are e.g. useful as hash values in unordered maps.
+  uint64_t NumericRepresentation() const { return id_; }
+
+ private:
+  explicit AudioCodecPairId(uint64_t id) : id_(id) {}
+
+  uint64_t id_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_decoder.cc

@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/audio_decoder.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+
+class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OldStyleEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
+      : decoder_(decoder), payload_(std::move(payload)) {}
+
+  size_t Duration() const override {
+    const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    return ret < 0 ? 0 : static_cast<size_t>(ret);
+  }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    auto speech_type = AudioDecoder::kSpeech;
+    const int ret = decoder_->Decode(
+        payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+        decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    return ret < 0 ? absl::nullopt
+                   : absl::optional<DecodeResult>(
+                         {static_cast<size_t>(ret), speech_type});
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+};
+
+}  // namespace
+
+bool AudioDecoder::EncodedAudioFrame::IsDtxPacket() const {
+  return false;
+}
+
+AudioDecoder::ParseResult::ParseResult() = default;
+AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
+AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
+                                       int priority,
+                                       std::unique_ptr<EncodedAudioFrame> frame)
+    : timestamp(timestamp), priority(priority), frame(std::move(frame)) {
+  RTC_DCHECK_GE(priority, 0);
+}
+
+AudioDecoder::ParseResult::~ParseResult() = default;
+
+AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
+    ParseResult&& b) = default;
+
+std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OldStyleEncodedFrame(this, std::move(payload)));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoder::Decode(const uint8_t* encoded,
+                         size_t encoded_len,
+                         int sample_rate_hz,
+                         size_t max_decoded_bytes,
+                         int16_t* decoded,
+                         SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
+  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
+  int duration = PacketDuration(encoded, encoded_len);
+  if (duration >= 0 &&
+      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
+    return -1;
+  }
+  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                        speech_type);
+}
+
+int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
+                                  size_t encoded_len,
+                                  int sample_rate_hz,
+                                  size_t max_decoded_bytes,
+                                  int16_t* decoded,
+                                  SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
+  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
+  int duration = PacketDurationRedundant(encoded, encoded_len);
+  if (duration >= 0 &&
+      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
+    return -1;
+  }
+  return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                                 speech_type);
+}
+
+int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
+                                          size_t encoded_len,
+                                          int sample_rate_hz,
+                                          int16_t* decoded,
+                                          SpeechType* speech_type) {
+  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                        speech_type);
+}
+
+bool AudioDecoder::HasDecodePlc() const {
+  return false;
+}
+
+size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return 0;
+}
+
+// TODO(bugs.webrtc.org/9676): Remove default implementation.
+void AudioDecoder::GeneratePlc(size_t /*requested_samples_per_channel*/,
+                               rtc::BufferT<int16_t>* /*concealment_audio*/) {}
+
+int AudioDecoder::ErrorCode() {
+  return 0;
+}
+
+int AudioDecoder::PacketDuration(const uint8_t* encoded,
+                                 size_t encoded_len) const {
+  return kNotImplemented;
+}
+
+int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
+                                          size_t encoded_len) const {
+  return kNotImplemented;
+}
+
+bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
+                                size_t encoded_len) const {
+  return false;
+}
+
+AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
+  switch (type) {
+    case 0:  // TODO(hlundin): Both iSAC and Opus return 0 for speech.
+    case 1:
+      return kSpeech;
+    case 2:
+      return kComfortNoise;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return kSpeech;
+  }
+}
+
+constexpr int AudioDecoder::kMaxNumberOfChannels;
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_decoder.h

@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_DECODER_H_
+#define API_AUDIO_CODECS_AUDIO_DECODER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoder {
+ public:
+  enum SpeechType {
+    kSpeech = 1,
+    kComfortNoise = 2,
+  };
+
+  // Used by PacketDuration below. Save the value -1 for errors.
+  enum { kNotImplemented = -2 };
+
+  AudioDecoder() = default;
+  virtual ~AudioDecoder() = default;
+
+  AudioDecoder(const AudioDecoder&) = delete;
+  AudioDecoder& operator=(const AudioDecoder&) = delete;
+
+  class EncodedAudioFrame {
+   public:
+    struct DecodeResult {
+      size_t num_decoded_samples;
+      SpeechType speech_type;
+    };
+
+    virtual ~EncodedAudioFrame() = default;
+
+    // Returns the duration in samples-per-channel of this audio frame.
+    // If no duration can be ascertained, returns zero.
+    virtual size_t Duration() const = 0;
+
+    // Returns true if this packet contains DTX.
+    virtual bool IsDtxPacket() const;
+
+    // Decodes this frame of audio and writes the result in `decoded`.
+    // `decoded` must be large enough to store as many samples as indicated by a
+    // call to Duration() . On success, returns an absl::optional containing the
+    // total number of samples across all channels, as well as whether the
+    // decoder produced comfort noise or speech. On failure, returns an empty
+    // absl::optional. Decode may be called at most once per frame object.
+    virtual absl::optional<DecodeResult> Decode(
+        rtc::ArrayView<int16_t> decoded) const = 0;
+  };
+
+  struct ParseResult {
+    ParseResult();
+    ParseResult(uint32_t timestamp,
+                int priority,
+                std::unique_ptr<EncodedAudioFrame> frame);
+    ParseResult(ParseResult&& b);
+    ~ParseResult();
+
+    ParseResult& operator=(ParseResult&& b);
+
+    // The timestamp of the frame is in samples per channel.
+    uint32_t timestamp;
+    // The relative priority of the frame compared to other frames of the same
+    // payload and the same timeframe. A higher value means a lower priority.
+    // The highest priority is zero - negative values are not allowed.
+    int priority;
+    std::unique_ptr<EncodedAudioFrame> frame;
+  };
+
+  // Let the decoder parse this payload and prepare zero or more decodable
+  // frames. Each frame must be between 10 ms and 120 ms long. The caller must
+  // ensure that the AudioDecoder object outlives any frame objects returned by
+  // this call. The decoder is free to swap or move the data from the `payload`
+  // buffer. `timestamp` is the input timestamp, in samples, corresponding to
+  // the start of the payload.
+  virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                                uint32_t timestamp);
+
+  // TODO(bugs.webrtc.org/10098): The Decode and DecodeRedundant methods are
+  // obsolete; callers should call ParsePayload instead. For now, subclasses
+  // must still implement DecodeInternal.
+
+  // Decodes `encode_len` bytes from `encoded` and writes the result in
+  // `decoded`. The maximum bytes allowed to be written into `decoded` is
+  // `max_decoded_bytes`. Returns the total number of samples across all
+  // channels. If the decoder produced comfort noise, `speech_type`
+  // is set to kComfortNoise, otherwise it is kSpeech. The desired output
+  // sample rate is provided in `sample_rate_hz`, which must be valid for the
+  // codec at hand.
+  int Decode(const uint8_t* encoded,
+             size_t encoded_len,
+             int sample_rate_hz,
+             size_t max_decoded_bytes,
+             int16_t* decoded,
+             SpeechType* speech_type);
+
+  // Same as Decode(), but interfaces to the decoders redundant decode function.
+  // The default implementation simply calls the regular Decode() method.
+  int DecodeRedundant(const uint8_t* encoded,
+                      size_t encoded_len,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded,
+                      SpeechType* speech_type);
+
+  // Indicates if the decoder implements the DecodePlc method.
+  virtual bool HasDecodePlc() const;
+
+  // Calls the packet-loss concealment of the decoder to update the state after
+  // one or several lost packets. The caller has to make sure that the
+  // memory allocated in `decoded` should accommodate `num_frames` frames.
+  virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
+
+  // Asks the decoder to generate packet-loss concealment and append it to the
+  // end of `concealment_audio`. The concealment audio should be in
+  // channel-interleaved format, with as many channels as the last decoded
+  // packet produced. The implementation must produce at least
+  // requested_samples_per_channel, or nothing at all. This is a signal to the
+  // caller to conceal the loss with other means. If the implementation provides
+  // concealment samples, it is also responsible for "stitching" it together
+  // with the decoded audio on either side of the concealment.
+  // Note: The default implementation of GeneratePlc will be deleted soon. All
+  // implementations must provide their own, which can be a simple as a no-op.
+  // TODO(bugs.webrtc.org/9676): Remove default implementation.
+  virtual void GeneratePlc(size_t requested_samples_per_channel,
+                           rtc::BufferT<int16_t>* concealment_audio);
+
+  // Resets the decoder state (empty buffers etc.).
+  virtual void Reset() = 0;
+
+  // Returns the last error code from the decoder.
+  virtual int ErrorCode();
+
+  // Returns the duration in samples-per-channel of the payload in `encoded`
+  // which is `encoded_len` bytes long. Returns kNotImplemented if no duration
+  // estimate is available, or -1 in case of an error.
+  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+
+  // Returns the duration in samples-per-channel of the redandant payload in
+  // `encoded` which is `encoded_len` bytes long. Returns kNotImplemented if no
+  // duration estimate is available, or -1 in case of an error.
+  virtual int PacketDurationRedundant(const uint8_t* encoded,
+                                      size_t encoded_len) const;
+
+  // Detects whether a packet has forward error correction. The packet is
+  // comprised of the samples in `encoded` which is `encoded_len` bytes long.
+  // Returns true if the packet has FEC and false otherwise.
+  virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
+
+  // Returns the actual sample rate of the decoder's output. This value may not
+  // change during the lifetime of the decoder.
+  virtual int SampleRateHz() const = 0;
+
+  // The number of channels in the decoder's output. This value may not change
+  // during the lifetime of the decoder.
+  virtual size_t Channels() const = 0;
+
+  // The maximum number of audio channels supported by WebRTC decoders.
+  static constexpr int kMaxNumberOfChannels = 24;
+
+ protected:
+  static SpeechType ConvertSpeechType(int16_t type);
+
+  virtual int DecodeInternal(const uint8_t* encoded,
+                             size_t encoded_len,
+                             int sample_rate_hz,
+                             int16_t* decoded,
+                             SpeechType* speech_type) = 0;
+
+  virtual int DecodeRedundantInternal(const uint8_t* encoded,
+                                      size_t encoded_len,
+                                      int sample_rate_hz,
+                                      int16_t* decoded,
+                                      SpeechType* speech_type);
+};
+
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_AUDIO_DECODER_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_decoder_factory.h

@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
+#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/ref_count.h"
+
+namespace webrtc {
+
+// A factory that creates AudioDecoders.
+class AudioDecoderFactory : public RefCountInterface {
+ public:
+  virtual std::vector<AudioCodecSpec> GetSupportedDecoders() = 0;
+
+  virtual bool IsSupportedDecoder(const SdpAudioFormat& format) = 0;
+
+  // Create a new decoder instance. The `codec_pair_id` argument is used to link
+  // encoders and decoders that talk to the same remote entity: if a
+  // AudioEncoderFactory::MakeAudioEncoder() and a
+  // AudioDecoderFactory::MakeAudioDecoder() call receive non-null IDs that
+  // compare equal, the factory implementations may assume that the encoder and
+  // decoder form a pair. (The intended use case for this is to set up
+  // communication between the AudioEncoder and AudioDecoder instances, which is
+  // needed for some codecs with built-in bandwidth adaptation.)
+  //
+  // Returns null if the format isn't supported.
+  //
+  // Note: Implementations need to be robust against combinations other than
+  // one encoder, one decoder getting the same ID; such decoders must still
+  // work.
+  virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_decoder_factory_template.h

@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
+#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/field_trials_view.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+namespace audio_decoder_factory_template_impl {
+
+template <typename... Ts>
+struct Helper;
+
+// Base case: 0 template parameters.
+template <>
+struct Helper<> {
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {}
+  static bool IsSupportedDecoder(const SdpAudioFormat& format) { return false; }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    return nullptr;
+  }
+};
+
+// Inductive case: Called with n + 1 template parameters; calls subroutines
+// with n template parameters.
+template <typename T, typename... Ts>
+struct Helper<T, Ts...> {
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    T::AppendSupportedDecoders(specs);
+    Helper<Ts...>::AppendSupportedDecoders(specs);
+  }
+  static bool IsSupportedDecoder(const SdpAudioFormat& format) {
+    auto opt_config = T::SdpToConfig(format);
+    static_assert(std::is_same<decltype(opt_config),
+                               absl::optional<typename T::Config>>::value,
+                  "T::SdpToConfig() must return a value of type "
+                  "absl::optional<T::Config>");
+    return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format);
+  }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    auto opt_config = T::SdpToConfig(format);
+    return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
+                      : Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id,
+                                                        field_trials);
+  }
+};
+
+template <typename... Ts>
+class AudioDecoderFactoryT : public AudioDecoderFactory {
+ public:
+  explicit AudioDecoderFactoryT(const FieldTrialsView* field_trials) {
+    field_trials_ = field_trials;
+  }
+
+  std::vector<AudioCodecSpec> GetSupportedDecoders() override {
+    std::vector<AudioCodecSpec> specs;
+    Helper<Ts...>::AppendSupportedDecoders(&specs);
+    return specs;
+  }
+
+  bool IsSupportedDecoder(const SdpAudioFormat& format) override {
+    return Helper<Ts...>::IsSupportedDecoder(format);
+  }
+
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) override {
+    return Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id,
+                                           field_trials_);
+  }
+
+  const FieldTrialsView* field_trials_;
+};
+
+}  // namespace audio_decoder_factory_template_impl
+
+// Make an AudioDecoderFactory that can create instances of the given decoders.
+//
+// Each decoder type is given as a template argument to the function; it should
+// be a struct with the following static member functions:
+//
+//   // Converts `audio_format` to a ConfigType instance. Returns an empty
+//   // optional if `audio_format` doesn't correctly specify a decoder of our
+//   // type.
+//   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+//
+//   // Appends zero or more AudioCodecSpecs to the list that will be returned
+//   // by AudioDecoderFactory::GetSupportedDecoders().
+//   void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+//
+//   // Creates an AudioDecoder for the specified format. Used to implement
+//   // AudioDecoderFactory::MakeAudioDecoder().
+//   std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+//       const ConfigType& config,
+//       absl::optional<AudioCodecPairId> codec_pair_id);
+//
+// ConfigType should be a type that encapsulates all the settings needed to
+// create an AudioDecoder. T::Config (where T is the decoder struct) should
+// either be the config type, or an alias for it.
+//
+// Whenever it tries to do something, the new factory will try each of the
+// decoder types in the order they were specified in the template argument
+// list, stopping at the first one that claims to be able to do the job.
+//
+// TODO(kwiberg): Point at CreateBuiltinAudioDecoderFactory() for an example of
+// how it is used.
+template <typename... Ts>
+rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory(
+    const FieldTrialsView* field_trials = nullptr) {
+  // There's no technical reason we couldn't allow zero template parameters,
+  // but such a factory couldn't create any decoders, and callers can do this
+  // by mistake by simply forgetting the <> altogether. So we forbid it in
+  // order to prevent caller foot-shooting.
+  static_assert(sizeof...(Ts) >= 1,
+                "Caller must give at least one template parameter");
+
+  return rtc::make_ref_counted<
+      audio_decoder_factory_template_impl::AudioDecoderFactoryT<Ts...>>(
+      field_trials);
+}
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_encoder.cc

@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/audio_encoder.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+ANAStats::ANAStats() = default;
+ANAStats::~ANAStats() = default;
+ANAStats::ANAStats(const ANAStats&) = default;
+
+AudioEncoder::EncodedInfo::EncodedInfo() = default;
+AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
+AudioEncoder::EncodedInfo::~EncodedInfo() = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
+    const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
+    default;
+
+int AudioEncoder::RtpTimestampRateHz() const {
+  return SampleRateHz();
+}
+
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
+  RTC_CHECK_EQ(audio.size(),
+               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
+
+  const size_t old_size = encoded->size();
+  EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
+  RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
+  return info;
+}
+
+bool AudioEncoder::SetFec(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::SetDtx(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::GetDtx() const {
+  return false;
+}
+
+bool AudioEncoder::SetApplication(Application application) {
+  return false;
+}
+
+void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoder::ReclaimContainedEncoders() {
+  return nullptr;
+}
+
+bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
+                                             RtcEventLog* event_log) {
+  return false;
+}
+
+void AudioEncoder::DisableAudioNetworkAdaptor() {}
+
+void AudioEncoder::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {}
+
+void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
+    float uplink_recoverable_packet_loss_fraction) {
+  RTC_DCHECK_NOTREACHED();
+}
+
+void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
+}
+
+void AudioEncoder::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {}
+
+void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
+                            update.bwe_period.ms());
+}
+
+void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
+
+void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
+
+void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                               int max_frame_length_ms) {}
+
+ANAStats AudioEncoder::GetANAStats() const {
+  return ANAStats();
+}
+
+constexpr int AudioEncoder::kMaxNumberOfChannels;
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_encoder.h

@@ -0,0 +1,269 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_H_
+#define API_AUDIO_CODECS_AUDIO_ENCODER_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/bitrate_allocation.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+// Statistics related to Audio Network Adaptation.
+struct ANAStats {
+  ANAStats();
+  ANAStats(const ANAStats&);
+  ~ANAStats();
+  // Number of actions taken by the ANA bitrate controller since the start of
+  // the call. If this value is not set, it indicates that the bitrate
+  // controller is disabled.
+  absl::optional<uint32_t> bitrate_action_counter;
+  // Number of actions taken by the ANA channel controller since the start of
+  // the call. If this value is not set, it indicates that the channel
+  // controller is disabled.
+  absl::optional<uint32_t> channel_action_counter;
+  // Number of actions taken by the ANA DTX controller since the start of the
+  // call. If this value is not set, it indicates that the DTX controller is
+  // disabled.
+  absl::optional<uint32_t> dtx_action_counter;
+  // Number of actions taken by the ANA FEC controller since the start of the
+  // call. If this value is not set, it indicates that the FEC controller is
+  // disabled.
+  absl::optional<uint32_t> fec_action_counter;
+  // Number of times the ANA frame length controller decided to increase the
+  // frame length since the start of the call. If this value is not set, it
+  // indicates that the frame length controller is disabled.
+  absl::optional<uint32_t> frame_length_increase_counter;
+  // Number of times the ANA frame length controller decided to decrease the
+  // frame length since the start of the call. If this value is not set, it
+  // indicates that the frame length controller is disabled.
+  absl::optional<uint32_t> frame_length_decrease_counter;
+  // The uplink packet loss fractions as set by the ANA FEC controller. If this
+  // value is not set, it indicates that the ANA FEC controller is not active.
+  absl::optional<float> uplink_packet_loss_fraction;
+};
+
+// This is the interface class for encoders in AudioCoding module. Each codec
+// type must have an implementation of this class.
+class AudioEncoder {
+ public:
+  // Used for UMA logging of codec usage. The same codecs, with the
+  // same values, must be listed in
+  // src/tools/metrics/histograms/histograms.xml in chromium to log
+  // correct values.
+  enum class CodecType {
+    kOther = 0,  // Codec not specified, and/or not listed in this enum
+    kOpus = 1,
+    kIsac = 2,
+    kPcmA = 3,
+    kPcmU = 4,
+    kG722 = 5,
+    kIlbc = 6,
+
+    // Number of histogram bins in the UMA logging of codec types. The
+    // total number of different codecs that are logged cannot exceed this
+    // number.
+    kMaxLoggedAudioCodecTypes
+  };
+
+  struct EncodedInfoLeaf {
+    size_t encoded_bytes = 0;
+    uint32_t encoded_timestamp = 0;
+    int payload_type = 0;
+    bool send_even_if_empty = false;
+    bool speech = true;
+    CodecType encoder_type = CodecType::kOther;
+  };
+
+  // This is the main struct for auxiliary encoding information. Each encoded
+  // packet should be accompanied by one EncodedInfo struct, containing the
+  // total number of `encoded_bytes`, the `encoded_timestamp` and the
+  // `payload_type`. If the packet contains redundant encodings, the `redundant`
+  // vector will be populated with EncodedInfoLeaf structs. Each struct in the
+  // vector represents one encoding; the order of structs in the vector is the
+  // same as the order in which the actual payloads are written to the byte
+  // stream. When EncoderInfoLeaf structs are present in the vector, the main
+  // struct's `encoded_bytes` will be the sum of all the `encoded_bytes` in the
+  // vector.
+  struct EncodedInfo : public EncodedInfoLeaf {
+    EncodedInfo();
+    EncodedInfo(const EncodedInfo&);
+    EncodedInfo(EncodedInfo&&);
+    ~EncodedInfo();
+    EncodedInfo& operator=(const EncodedInfo&);
+    EncodedInfo& operator=(EncodedInfo&&);
+
+    std::vector<EncodedInfoLeaf> redundant;
+  };
+
+  virtual ~AudioEncoder() = default;
+
+  // Returns the input sample rate in Hz and the number of input channels.
+  // These are constants set at instantiation time.
+  virtual int SampleRateHz() const = 0;
+  virtual size_t NumChannels() const = 0;
+
+  // Returns the rate at which the RTP timestamps are updated. The default
+  // implementation returns SampleRateHz().
+  virtual int RtpTimestampRateHz() const;
+
+  // Returns the number of 10 ms frames the encoder will put in the next
+  // packet. This value may only change when Encode() outputs a packet; i.e.,
+  // the encoder may vary the number of 10 ms frames from packet to packet, but
+  // it must decide the length of the next packet no later than when outputting
+  // the preceding packet.
+  virtual size_t Num10MsFramesInNextPacket() const = 0;
+
+  // Returns the maximum value that can be returned by
+  // Num10MsFramesInNextPacket().
+  virtual size_t Max10MsFramesInAPacket() const = 0;
+
+  // Returns the current target bitrate in bits/s. The value -1 means that the
+  // codec adapts the target automatically, and a current target cannot be
+  // provided.
+  virtual int GetTargetBitrate() const = 0;
+
+  // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
+  // NumChannels() samples). Multi-channel audio must be sample-interleaved.
+  // The encoder appends zero or more bytes of output to `encoded` and returns
+  // additional encoding information.  Encode() checks some preconditions, calls
+  // EncodeImpl() which does the actual work, and then checks some
+  // postconditions.
+  EncodedInfo Encode(uint32_t rtp_timestamp,
+                     rtc::ArrayView<const int16_t> audio,
+                     rtc::Buffer* encoded);
+
+  // Resets the encoder to its starting state, discarding any input that has
+  // been fed to the encoder but not yet emitted in a packet.
+  virtual void Reset() = 0;
+
+  // Enables or disables codec-internal FEC (forward error correction). Returns
+  // true if the codec was able to comply. The default implementation returns
+  // true when asked to disable FEC and false when asked to enable it (meaning
+  // that FEC isn't supported).
+  virtual bool SetFec(bool enable);
+
+  // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
+  // able to comply. The default implementation returns true when asked to
+  // disable DTX and false when asked to enable it (meaning that DTX isn't
+  // supported).
+  virtual bool SetDtx(bool enable);
+
+  // Returns the status of codec-internal DTX. The default implementation always
+  // returns false.
+  virtual bool GetDtx() const;
+
+  // Sets the application mode. Returns true if the codec was able to comply.
+  // The default implementation just returns false.
+  enum class Application { kSpeech, kAudio };
+  virtual bool SetApplication(Application application);
+
+  // Tells the encoder about the highest sample rate the decoder is expected to
+  // use when decoding the bitstream. The encoder would typically use this
+  // information to adjust the quality of the encoding. The default
+  // implementation does nothing.
+  virtual void SetMaxPlaybackRate(int frequency_hz);
+
+  // Tells the encoder what average bitrate we'd like it to produce. The
+  // encoder is free to adjust or disregard the given bitrate (the default
+  // implementation does the latter).
+  ABSL_DEPRECATED("Use OnReceivedTargetAudioBitrate instead")
+  virtual void SetTargetBitrate(int target_bps);
+
+  // Causes this encoder to let go of any other encoders it contains, and
+  // returns a pointer to an array where they are stored (which is required to
+  // live as long as this encoder). Unless the returned array is empty, you may
+  // not call any methods on this encoder afterwards, except for the
+  // destructor. The default implementation just returns an empty array.
+  // NOTE: This method is subject to change. Do not call or override it.
+  virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+  ReclaimContainedEncoders();
+
+  // Enables audio network adaptor. Returns true if successful.
+  virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                         RtcEventLog* event_log);
+
+  // Disables audio network adaptor.
+  virtual void DisableAudioNetworkAdaptor();
+
+  // Provides uplink packet loss fraction to this encoder to allow it to adapt.
+  // `uplink_packet_loss_fraction` is in the range [0.0, 1.0].
+  virtual void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction);
+
+  ABSL_DEPRECATED("")
+  virtual void OnReceivedUplinkRecoverablePacketLossFraction(
+      float uplink_recoverable_packet_loss_fraction);
+
+  // Provides target audio bitrate to this encoder to allow it to adapt.
+  virtual void OnReceivedTargetAudioBitrate(int target_bps);
+
+  // Provides target audio bitrate and corresponding probing interval of
+  // the bandwidth estimator to this encoder to allow it to adapt.
+  virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
+                                         absl::optional<int64_t> bwe_period_ms);
+
+  // Provides target audio bitrate and corresponding probing interval of
+  // the bandwidth estimator to this encoder to allow it to adapt.
+  virtual void OnReceivedUplinkAllocation(BitrateAllocationUpdate update);
+
+  // Provides RTT to this encoder to allow it to adapt.
+  virtual void OnReceivedRtt(int rtt_ms);
+
+  // Provides overhead to this encoder to adapt. The overhead is the number of
+  // bytes that will be added to each packet the encoder generates.
+  virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
+
+  // To allow encoder to adapt its frame length, it must be provided the frame
+  // length range that receivers can accept.
+  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                           int max_frame_length_ms);
+
+  // Get statistics related to audio network adaptation.
+  virtual ANAStats GetANAStats() const;
+
+  // The range of frame lengths that are supported or nullopt if there's no such
+  // information. This is used together with the bitrate range to calculate the
+  // full bitrate range, including overhead.
+  virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const = 0;
+
+  // The range of payload bitrates that are supported. This is used together
+  // with the frame length range to calculate the full bitrate range, including
+  // overhead.
+  virtual absl::optional<std::pair<DataRate, DataRate>> GetBitrateRange()
+      const {
+    return absl::nullopt;
+  }
+
+  // The maximum number of audio channels supported by WebRTC encoders.
+  static constexpr int kMaxNumberOfChannels = 24;
+
+ protected:
+  // Subclasses implement this to perform the actual encoding. Called by
+  // Encode().
+  virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                                 rtc::ArrayView<const int16_t> audio,
+                                 rtc::Buffer* encoded) = 0;
+};
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_AUDIO_ENCODER_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_encoder_factory.h

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
+#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// A factory that creates AudioEncoders.
+class AudioEncoderFactory : public rtc::RefCountInterface {
+ public:
+  // Returns a prioritized list of audio codecs, to use for signaling etc.
+  virtual std::vector<AudioCodecSpec> GetSupportedEncoders() = 0;
+
+  // Returns information about how this format would be encoded, provided it's
+  // supported. More format and format variations may be supported than those
+  // returned by GetSupportedEncoders().
+  virtual absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) = 0;
+
+  // Creates an AudioEncoder for the specified format. The encoder will tags its
+  // payloads with the specified payload type. The `codec_pair_id` argument is
+  // used to link encoders and decoders that talk to the same remote entity: if
+  // a AudioEncoderFactory::MakeAudioEncoder() and a
+  // AudioDecoderFactory::MakeAudioDecoder() call receive non-null IDs that
+  // compare equal, the factory implementations may assume that the encoder and
+  // decoder form a pair. (The intended use case for this is to set up
+  // communication between the AudioEncoder and AudioDecoder instances, which is
+  // needed for some codecs with built-in bandwidth adaptation.)
+  //
+  // Returns null if the format isn't supported.
+  //
+  // Note: Implementations need to be robust against combinations other than
+  // one encoder, one decoder getting the same ID; such encoders must still
+  // work.
+  //
+  // TODO(ossu): Try to avoid audio encoders having to know their payload type.
+  virtual std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_encoder_factory_template.h

@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
+#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/field_trials_view.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+namespace audio_encoder_factory_template_impl {
+
+template <typename... Ts>
+struct Helper;
+
+// Base case: 0 template parameters.
+template <>
+struct Helper<> {
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {}
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) {
+    return absl::nullopt;
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    return nullptr;
+  }
+};
+
+// Inductive case: Called with n + 1 template parameters; calls subroutines
+// with n template parameters.
+template <typename T, typename... Ts>
+struct Helper<T, Ts...> {
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    T::AppendSupportedEncoders(specs);
+    Helper<Ts...>::AppendSupportedEncoders(specs);
+  }
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) {
+    auto opt_config = T::SdpToConfig(format);
+    static_assert(std::is_same<decltype(opt_config),
+                               absl::optional<typename T::Config>>::value,
+                  "T::SdpToConfig() must return a value of type "
+                  "absl::optional<T::Config>");
+    return opt_config ? absl::optional<AudioCodecInfo>(
+                            T::QueryAudioEncoder(*opt_config))
+                      : Helper<Ts...>::QueryAudioEncoder(format);
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    auto opt_config = T::SdpToConfig(format);
+    if (opt_config) {
+      return T::MakeAudioEncoder(*opt_config, payload_type, codec_pair_id);
+    } else {
+      return Helper<Ts...>::MakeAudioEncoder(payload_type, format,
+                                             codec_pair_id, field_trials);
+    }
+  }
+};
+
+template <typename... Ts>
+class AudioEncoderFactoryT : public AudioEncoderFactory {
+ public:
+  explicit AudioEncoderFactoryT(const FieldTrialsView* field_trials) {
+    field_trials_ = field_trials;
+  }
+
+  std::vector<AudioCodecSpec> GetSupportedEncoders() override {
+    std::vector<AudioCodecSpec> specs;
+    Helper<Ts...>::AppendSupportedEncoders(&specs);
+    return specs;
+  }
+
+  absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) override {
+    return Helper<Ts...>::QueryAudioEncoder(format);
+  }
+
+  std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) override {
+    return Helper<Ts...>::MakeAudioEncoder(payload_type, format, codec_pair_id,
+                                           field_trials_);
+  }
+
+  const FieldTrialsView* field_trials_;
+};
+
+}  // namespace audio_encoder_factory_template_impl
+
+// Make an AudioEncoderFactory that can create instances of the given encoders.
+//
+// Each encoder type is given as a template argument to the function; it should
+// be a struct with the following static member functions:
+//
+//   // Converts `audio_format` to a ConfigType instance. Returns an empty
+//   // optional if `audio_format` doesn't correctly specify an encoder of our
+//   // type.
+//   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+//
+//   // Appends zero or more AudioCodecSpecs to the list that will be returned
+//   // by AudioEncoderFactory::GetSupportedEncoders().
+//   void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+//
+//   // Returns information about how this format would be encoded. Used to
+//   // implement AudioEncoderFactory::QueryAudioEncoder().
+//   AudioCodecInfo QueryAudioEncoder(const ConfigType& config);
+//
+//   // Creates an AudioEncoder for the specified format. Used to implement
+//   // AudioEncoderFactory::MakeAudioEncoder().
+//   std::unique_ptr<AudioDecoder> MakeAudioEncoder(
+//       const ConfigType& config,
+//       int payload_type,
+//       absl::optional<AudioCodecPairId> codec_pair_id);
+//
+// ConfigType should be a type that encapsulates all the settings needed to
+// create an AudioEncoder. T::Config (where T is the encoder struct) should
+// either be the config type, or an alias for it.
+//
+// Whenever it tries to do something, the new factory will try each of the
+// encoders in the order they were specified in the template argument list,
+// stopping at the first one that claims to be able to do the job.
+//
+// TODO(kwiberg): Point at CreateBuiltinAudioEncoderFactory() for an example of
+// how it is used.
+template <typename... Ts>
+rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory(
+    const FieldTrialsView* field_trials = nullptr) {
+  // There's no technical reason we couldn't allow zero template parameters,
+  // but such a factory couldn't create any encoders, and callers can do this
+  // by mistake by simply forgetting the <> altogether. So we forbid it in
+  // order to prevent caller foot-shooting.
+  static_assert(sizeof...(Ts) >= 1,
+                "Caller must give at least one template parameter");
+
+  return rtc::make_ref_counted<
+      audio_encoder_factory_template_impl::AudioEncoderFactoryT<Ts...>>(
+      field_trials);
+}
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_format.cc

@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/audio_format.h"
+
+#include <utility>
+
+#include "absl/strings/match.h"
+
+namespace webrtc {
+
+SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
+SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels)
+    : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels,
+                               const CodecParameterMap& param)
+    : name(name),
+      clockrate_hz(clockrate_hz),
+      num_channels(num_channels),
+      parameters(param) {}
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels,
+                               CodecParameterMap&& param)
+    : name(name),
+      clockrate_hz(clockrate_hz),
+      num_channels(num_channels),
+      parameters(std::move(param)) {}
+
+bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
+  return absl::EqualsIgnoreCase(name, o.name) &&
+         clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
+}
+
+SdpAudioFormat::~SdpAudioFormat() = default;
+SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
+SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
+
+bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+  return absl::EqualsIgnoreCase(a.name, b.name) &&
+         a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
+         a.parameters == b.parameters;
+}
+
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t num_channels,
+                               int bitrate_bps)
+    : AudioCodecInfo(sample_rate_hz,
+                     num_channels,
+                     bitrate_bps,
+                     bitrate_bps,
+                     bitrate_bps) {}
+
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t num_channels,
+                               int default_bitrate_bps,
+                               int min_bitrate_bps,
+                               int max_bitrate_bps)
+    : sample_rate_hz(sample_rate_hz),
+      num_channels(num_channels),
+      default_bitrate_bps(default_bitrate_bps),
+      min_bitrate_bps(min_bitrate_bps),
+      max_bitrate_bps(max_bitrate_bps) {
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  RTC_DCHECK_GT(num_channels, 0);
+  RTC_DCHECK_GE(min_bitrate_bps, 0);
+  RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
+  RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/audio_format.h

@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_AUDIO_FORMAT_H_
+#define API_AUDIO_CODECS_AUDIO_FORMAT_H_
+
+#include <stddef.h>
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/rtp_parameters.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// SDP specification for a single audio codec.
+struct RTC_EXPORT SdpAudioFormat {
+  using Parameters [[deprecated(("Use webrtc::CodecParameterMap"))]] =
+      std::map<std::string, std::string>;
+
+  SdpAudioFormat(const SdpAudioFormat&);
+  SdpAudioFormat(SdpAudioFormat&&);
+  SdpAudioFormat(absl::string_view name, int clockrate_hz, size_t num_channels);
+  SdpAudioFormat(absl::string_view name,
+                 int clockrate_hz,
+                 size_t num_channels,
+                 const CodecParameterMap& param);
+  SdpAudioFormat(absl::string_view name,
+                 int clockrate_hz,
+                 size_t num_channels,
+                 CodecParameterMap&& param);
+  ~SdpAudioFormat();
+
+  // Returns true if this format is compatible with `o`. In SDP terminology:
+  // would it represent the same codec between an offer and an answer? As
+  // opposed to operator==, this method disregards codec parameters.
+  bool Matches(const SdpAudioFormat& o) const;
+
+  SdpAudioFormat& operator=(const SdpAudioFormat&);
+  SdpAudioFormat& operator=(SdpAudioFormat&&);
+
+  friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b);
+  friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+    return !(a == b);
+  }
+
+  std::string name;
+  int clockrate_hz;
+  size_t num_channels;
+  CodecParameterMap parameters;
+};
+
+// Information about how an audio format is treated by the codec implementation.
+// Contains basic information, such as sample rate and number of channels, which
+// isn't uniformly presented by SDP. Also contains flags indicating support for
+// integrating with other parts of WebRTC, like external VAD and comfort noise
+// level calculation.
+//
+// To avoid API breakage, and make the code clearer, AudioCodecInfo should not
+// be directly initializable with any flags indicating optional support. If it
+// were, these initializers would break any time a new flag was added. It's also
+// more difficult to understand:
+//   AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true};
+// than
+//   AudioCodecInfo info(16000, 1, 32000);
+//   info.allow_comfort_noise = true;
+//   info.future_flag_b = true;
+//   info.future_flag_c = true;
+struct AudioCodecInfo {
+  AudioCodecInfo(int sample_rate_hz, size_t num_channels, int bitrate_bps);
+  AudioCodecInfo(int sample_rate_hz,
+                 size_t num_channels,
+                 int default_bitrate_bps,
+                 int min_bitrate_bps,
+                 int max_bitrate_bps);
+  AudioCodecInfo(const AudioCodecInfo& b) = default;
+  ~AudioCodecInfo() = default;
+
+  bool operator==(const AudioCodecInfo& b) const {
+    return sample_rate_hz == b.sample_rate_hz &&
+           num_channels == b.num_channels &&
+           default_bitrate_bps == b.default_bitrate_bps &&
+           min_bitrate_bps == b.min_bitrate_bps &&
+           max_bitrate_bps == b.max_bitrate_bps &&
+           allow_comfort_noise == b.allow_comfort_noise &&
+           supports_network_adaption == b.supports_network_adaption;
+  }
+
+  bool operator!=(const AudioCodecInfo& b) const { return !(*this == b); }
+
+  bool HasFixedBitrate() const {
+    RTC_DCHECK_GE(min_bitrate_bps, 0);
+    RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
+    RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
+    return min_bitrate_bps == max_bitrate_bps;
+  }
+
+  int sample_rate_hz;
+  size_t num_channels;
+  int default_bitrate_bps;
+  int min_bitrate_bps;
+  int max_bitrate_bps;
+
+  bool allow_comfort_noise = true;  // This codec can be used with an external
+                                    // comfort noise generator.
+  bool supports_network_adaption = false;  // This codec can adapt to varying
+                                           // network conditions.
+};
+
+// AudioCodecSpec ties an audio format to specific information about the codec
+// and its implementation.
+struct AudioCodecSpec {
+  bool operator==(const AudioCodecSpec& b) const {
+    return format == b.format && info == b.info;
+  }
+
+  bool operator!=(const AudioCodecSpec& b) const { return !(*this == b); }
+
+  SdpAudioFormat format;
+  AudioCodecInfo info;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_FORMAT_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc

@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/L16/audio_decoder_L16.h"
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/g711/audio_decoder_g711.h"
+#include "api/audio_codecs/g722/audio_decoder_g722.h"
+#if WEBRTC_USE_BUILTIN_ILBC
+#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"  // nogncheck
+#endif
+#if WEBRTC_USE_BUILTIN_OPUS
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"  // nogncheck
+#endif
+
+namespace webrtc {
+
+namespace {
+
+// Modify an audio decoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+    return T::MakeAudioDecoder(config, codec_pair_id);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
+  return CreateAudioDecoderFactory<
+
+#if WEBRTC_USE_BUILTIN_OPUS
+      AudioDecoderOpus, NotAdvertised<AudioDecoderMultiChannelOpus>,
+#endif
+
+      AudioDecoderG722,
+
+#if WEBRTC_USE_BUILTIN_ILBC
+      AudioDecoderIlbc,
+#endif
+
+      AudioDecoderG711, NotAdvertised<AudioDecoderL16>>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/builtin_audio_decoder_factory.h

@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+#define API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create the built-in types of audio decoders.
+// Note: This will link with all the code implementing those codecs, so if you
+// only need a subset of the codecs, consider using
+// CreateAudioDecoderFactory<...codecs listed here...>() or
+// CreateOpusAudioDecoderFactory() instead.
+rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/builtin_audio_encoder_factory.cc

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+#if WEBRTC_USE_BUILTIN_ILBC
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"  // nogncheck
+#endif
+#if WEBRTC_USE_BUILTIN_OPUS
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"  // nogncheck
+#endif
+
+namespace webrtc {
+
+namespace {
+
+// Modify an audio encoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static AudioCodecInfo QueryAudioEncoder(const Config& config) {
+    return T::QueryAudioEncoder(config);
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr) {
+    return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
+                               field_trials);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() {
+  return CreateAudioEncoderFactory<
+
+#if WEBRTC_USE_BUILTIN_OPUS
+      AudioEncoderOpus, NotAdvertised<AudioEncoderMultiChannelOpus>,
+#endif
+
+      AudioEncoderG722,
+
+#if WEBRTC_USE_BUILTIN_ILBC
+      AudioEncoderIlbc,
+#endif
+
+      AudioEncoderG711, NotAdvertised<AudioEncoderL16>>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/builtin_audio_encoder_factory.h

@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
+#define API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create the built-in types of audio encoders.
+// Note: This will link with all the code implementing those codecs, so if you
+// only need a subset of the codecs, consider using
+// CreateAudioEncoderFactory<...codecs listed here...>() or
+// CreateOpusAudioEncoderFactory() instead.
+rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/BUILD.gn

@@ -0,0 +1,55 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_g711") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_g711.cc",
+    "audio_encoder_g711.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_g711") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_g711.cc",
+    "audio_decoder_g711.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc

@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/g711/audio_decoder_g711.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
+  if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
+      (is_pcmu || is_pcma)) {
+    Config config;
+    config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
+    config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioDecoderG711::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  for (const char* type : {"PCMU", "PCMA"}) {
+    specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
+  }
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
+    const Config& config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.type) {
+    case Config::Type::kPcmU:
+      return std::make_unique<AudioDecoderPcmU>(config.num_channels);
+    case Config::Type::kPcmA:
+      return std::make_unique<AudioDecoderPcmA>(config.num_channels);
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_decoder_g711.h

@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
+#define API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G711 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderG711 {
+  struct Config {
+    enum class Type { kPcmU, kPcmA };
+    bool IsOk() const {
+      return (type == Type::kPcmU || type == Type::kPcmA) &&
+             num_channels >= 1 &&
+             num_channels <= AudioDecoder::kMaxNumberOfChannels;
+    }
+    Type type;
+    int num_channels;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc

@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
+  if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
+      (is_pcmu || is_pcma)) {
+    Config config;
+    config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
+    config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+    config.frame_size_ms = 20;
+    auto ptime_iter = format.parameters.find("ptime");
+    if (ptime_iter != format.parameters.end()) {
+      const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+      if (ptime && *ptime > 0) {
+        config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+      }
+    }
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioEncoderG711::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  for (const char* type : {"PCMU", "PCMA"}) {
+    specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
+  }
+}
+
+AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {8000, rtc::dchecked_cast<size_t>(config.num_channels),
+          64000 * config.num_channels};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
+    const Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.type) {
+    case Config::Type::kPcmU: {
+      AudioEncoderPcmU::Config impl_config;
+      impl_config.num_channels = config.num_channels;
+      impl_config.frame_size_ms = config.frame_size_ms;
+      impl_config.payload_type = payload_type;
+      return std::make_unique<AudioEncoderPcmU>(impl_config);
+    }
+    case Config::Type::kPcmA: {
+      AudioEncoderPcmA::Config impl_config;
+      impl_config.num_channels = config.num_channels;
+      impl_config.frame_size_ms = config.frame_size_ms;
+      impl_config.payload_type = payload_type;
+      return std::make_unique<AudioEncoderPcmA>(impl_config);
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+    }
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g711/audio_encoder_g711.h

@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
+#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G711 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderG711 {
+  struct Config {
+    enum class Type { kPcmU, kPcmA };
+    bool IsOk() const {
+      return (type == Type::kPcmU || type == Type::kPcmA) &&
+             frame_size_ms > 0 && frame_size_ms % 10 == 0 &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels;
+    }
+    Type type = Type::kPcmU;
+    int num_channels = 1;
+    int frame_size_ms = 20;
+  };
+  static absl::optional<AudioEncoderG711::Config> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/BUILD.gn

@@ -0,0 +1,62 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_source_set("audio_encoder_g722_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_encoder_g722_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
+}
+
+rtc_library("audio_encoder_g722") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_g722.cc",
+    "audio_encoder_g722.h",
+  ]
+  deps = [
+    ":audio_encoder_g722_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g722",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_g722") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_g722.cc",
+    "audio_decoder_g722.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g722",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc

@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/g722/audio_decoder_g722.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "G722") &&
+      format.clockrate_hz == 8000 &&
+      (format.num_channels == 1 || format.num_channels == 2)) {
+    return Config{rtc::dchecked_cast<int>(format.num_channels)};
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderG722::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  specs->push_back({{"G722", 8000, 1}, {16000, 1, 64000}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.num_channels) {
+    case 1:
+      return std::make_unique<AudioDecoderG722Impl>();
+    case 2:
+      return std::make_unique<AudioDecoderG722StereoImpl>();
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_decoder_g722.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
+#define API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G722 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderG722 {
+  struct Config {
+    bool IsOk() const { return num_channels == 1 || num_channels == 2; }
+    int num_channels;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "g722") ||
+      format.clockrate_hz != 8000) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderG722Config config;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderG722::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"G722", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
+    const AudioEncoderG722Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
+          64000 * config.num_channels};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
+    const AudioEncoderG722Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_encoder_g722.h

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G722 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderG722 {
+  using Config = AudioEncoderG722Config;
+  static absl::optional<AudioEncoderG722Config> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderG722Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderG722Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h

@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
+#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
+
+#include "api/audio_codecs/audio_encoder.h"
+
+namespace webrtc {
+
+struct AudioEncoderG722Config {
+  bool IsOk() const {
+    return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1 &&
+           num_channels <= AudioEncoder::kMaxNumberOfChannels;
+  }
+  int frame_size_ms = 20;
+  int num_channels = 1;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/BUILD.gn

@@ -0,0 +1,58 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_source_set("audio_encoder_ilbc_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_encoder_ilbc_config.h" ]
+}
+
+rtc_library("audio_encoder_ilbc") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_ilbc.cc",
+    "audio_encoder_ilbc.h",
+  ]
+  deps = [
+    ":audio_encoder_ilbc_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:ilbc",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_ilbc") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_ilbc.cc",
+    "audio_decoder_ilbc.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:ilbc",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "ILBC") &&
+      format.clockrate_hz == 8000 && format.num_channels == 1) {
+    return Config();
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderIlbc::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  specs->push_back({{"ILBC", 8000, 1}, {8000, 1, 13300}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return std::make_unique<AudioDecoderIlbcImpl>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h

@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+
+namespace webrtc {
+
+// ILBC decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct AudioDecoderIlbc {
+  struct Config {};  // Empty---no config values needed!
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc

@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+namespace {
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+}  // namespace
+
+absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
+      format.clockrate_hz != 8000 || format.num_channels != 1) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderIlbcConfig config;
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderIlbc::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"ILBC", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
+    const AudioEncoderIlbcConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return {8000, 1, GetIlbcBitrate(config.frame_size_ms)};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
+    const AudioEncoderIlbcConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderIlbcImpl>(config, payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/field_trials_view.h"
+
+namespace webrtc {
+
+// ILBC encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct AudioEncoderIlbc {
+  using Config = AudioEncoderIlbcConfig;
+  static absl::optional<AudioEncoderIlbcConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderIlbcConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderIlbcConfig& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h

@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
+
+namespace webrtc {
+
+struct AudioEncoderIlbcConfig {
+  bool IsOk() const {
+    return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
+            frame_size_ms == 60);
+  }
+  int frame_size_ms = 30;  // Valid values are 20, 30, 40, and 60 ms.
+  // Note that frame size 40 ms produces encodings with two 20 ms frames in
+  // them, and frame size 60 ms consists of two 30 ms frames.
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/BUILD.gn

@@ -0,0 +1,110 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_opus_config") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_encoder_multi_channel_opus_config.cc",
+    "audio_encoder_multi_channel_opus_config.h",
+    "audio_encoder_opus_config.cc",
+    "audio_encoder_opus_config.h",
+  ]
+  deps = [ "../../../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  defines = []
+  if (rtc_opus_variable_complexity) {
+    defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
+  } else {
+    defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
+  }
+}
+
+rtc_source_set("audio_decoder_opus_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_decoder_multi_channel_opus_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
+}
+
+rtc_library("audio_encoder_opus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  public = [ "audio_encoder_opus.h" ]
+  sources = [ "audio_encoder_opus.cc" ]
+  deps = [
+    ":audio_encoder_opus_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_opus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_opus.cc",
+    "audio_decoder_opus.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_encoder_multiopus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  public = [ "audio_encoder_multi_channel_opus.h" ]
+  sources = [ "audio_encoder_multi_channel_opus.cc" ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_multiopus",
+    "../../../rtc_base/system:rtc_export",
+    "../opus:audio_encoder_opus_config",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_decoder_multiopus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_multi_channel_opus.cc",
+    "audio_decoder_multi_channel_opus.h",
+  ]
+  deps = [
+    ":audio_decoder_opus_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_multiopus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc

@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderMultiChannelOpusConfig>
+AudioDecoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
+  return AudioDecoderMultiChannelOpusImpl::SdpToConfig(format);
+}
+
+void AudioDecoderMultiChannelOpus::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  // To get full utilization of the surround support of the Opus lib, we can
+  // mark which channel is the low frequency effects (LFE). But that is not done
+  // ATM.
+  {
+    AudioCodecInfo surround_5_1_opus_info{48000, 6,
+                                          /* default_bitrate_bps= */ 128000};
+    surround_5_1_opus_info.allow_comfort_noise = false;
+    surround_5_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                6,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,4,1,2,3,5"},
+                                 {"num_streams", "4"},
+                                 {"coupled_streams", "2"}}});
+    specs->push_back({std::move(opus_format), surround_5_1_opus_info});
+  }
+  {
+    AudioCodecInfo surround_7_1_opus_info{48000, 8,
+                                          /* default_bitrate_bps= */ 200000};
+    surround_7_1_opus_info.allow_comfort_noise = false;
+    surround_7_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                8,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,6,1,2,3,4,5,7"},
+                                 {"num_streams", "5"},
+                                 {"coupled_streams", "3"}}});
+    specs->push_back({std::move(opus_format), surround_7_1_opus_info});
+  }
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder(
+    AudioDecoderMultiChannelOpusConfig config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config);
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderMultiChannelOpus {
+  using Config = AudioDecoderMultiChannelOpusConfig;
+  static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      AudioDecoderMultiChannelOpusConfig config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+
+namespace webrtc {
+struct AudioDecoderMultiChannelOpusConfig {
+  // The number of channels that the decoder will output.
+  int num_channels;
+
+  // Number of mono or stereo encoded Opus streams.
+  int num_streams;
+
+  // Number of channel pairs coupled together, see RFC 7845 section
+  // 5.1.1. Has to be less than the number of streams.
+  int coupled_streams;
+
+  // Channel mapping table, defines the mapping from encoded streams to output
+  // channels. See RFC 7845 section 5.1.1.
+  std::vector<unsigned char> channel_mapping;
+
+  bool IsOk() const {
+    if (num_channels < 1 || num_channels > AudioDecoder::kMaxNumberOfChannels ||
+        num_streams < 0 || coupled_streams < 0) {
+      return false;
+    }
+    if (num_streams < coupled_streams) {
+      return false;
+    }
+    if (channel_mapping.size() != static_cast<size_t>(num_channels)) {
+      return false;
+    }
+
+    // Every mono stream codes one channel, every coupled stream codes two. This
+    // is the total coded channel count:
+    const int max_coded_channel = num_streams + coupled_streams;
+    for (const auto& x : channel_mapping) {
+      // Coded channels >= max_coded_channel don't exist. Except for 255, which
+      // tells Opus to put silence in output channel x.
+      if (x >= max_coded_channel && x != 255) {
+        return false;
+      }
+    }
+
+    if (num_channels > 255 || max_coded_channel >= 255) {
+      return false;
+    }
+    return true;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  //  API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc

@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+namespace webrtc {
+
+bool AudioDecoderOpus::Config::IsOk() const {
+  if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels != 1 && num_channels != 2) {
+    return false;
+  }
+  return true;
+}
+
+absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const auto num_channels = [&]() -> absl::optional<int> {
+    auto stereo = format.parameters.find("stereo");
+    if (stereo != format.parameters.end()) {
+      if (stereo->second == "0") {
+        return 1;
+      } else if (stereo->second == "1") {
+        return 2;
+      } else {
+        return absl::nullopt;  // Bad stereo parameter.
+      }
+    }
+    return 1;  // Default to mono.
+  }();
+  if (absl::EqualsIgnoreCase(format.name, "opus") &&
+      format.clockrate_hz == 48000 && format.num_channels == 2 &&
+      num_channels) {
+    Config config;
+    config.num_channels = *num_channels;
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioDecoderOpus::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
+  opus_info.allow_comfort_noise = false;
+  opus_info.supports_network_adaption = true;
+  SdpAudioFormat opus_format(
+      {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
+  specs->push_back({std::move(opus_format), opus_info});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderOpusImpl>(config.num_channels,
+                                                config.sample_rate_hz);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_decoder_opus.h

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderOpus {
+  struct Config {
+    bool IsOk() const;  // Checks if the values are currently OK.
+    int sample_rate_hz = 48000;
+    int num_channels = 1;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderMultiChannelOpusConfig>
+AudioEncoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
+  return AudioEncoderMultiChannelOpusImpl::SdpToConfig(format);
+}
+
+void AudioEncoderMultiChannelOpus::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  // To get full utilization of the surround support of the Opus lib, we can
+  // mark which channel is the low frequency effects (LFE). But that is not done
+  // ATM.
+  {
+    AudioCodecInfo surround_5_1_opus_info{48000, 6,
+                                          /* default_bitrate_bps= */ 128000};
+    surround_5_1_opus_info.allow_comfort_noise = false;
+    surround_5_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                6,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,4,1,2,3,5"},
+                                 {"num_streams", "4"},
+                                 {"coupled_streams", "2"}}});
+    specs->push_back({std::move(opus_format), surround_5_1_opus_info});
+  }
+  {
+    AudioCodecInfo surround_7_1_opus_info{48000, 8,
+                                          /* default_bitrate_bps= */ 200000};
+    surround_7_1_opus_info.allow_comfort_noise = false;
+    surround_7_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                8,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,6,1,2,3,4,5,7"},
+                                 {"num_streams", "5"},
+                                 {"coupled_streams", "3"}}});
+    specs->push_back({std::move(opus_format), surround_7_1_opus_info});
+  }
+}
+
+AudioCodecInfo AudioEncoderMultiChannelOpus::QueryAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  return AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder(config);
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config,
+                                                            payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderMultiChannelOpus {
+  using Config = AudioEncoderMultiChannelOpusConfig;
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc

@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kDefaultComplexity = 9;
+}  // namespace
+
+AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig()
+    : frame_size_ms(kDefaultFrameSizeMs),
+      num_channels(1),
+      application(ApplicationMode::kVoip),
+      bitrate_bps(32000),
+      fec_enabled(false),
+      cbr_enabled(false),
+      dtx_enabled(false),
+      max_playback_rate_hz(48000),
+      complexity(kDefaultComplexity),
+      num_streams(-1),
+      coupled_streams(-1) {}
+AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig(
+    const AudioEncoderMultiChannelOpusConfig&) = default;
+AudioEncoderMultiChannelOpusConfig::~AudioEncoderMultiChannelOpusConfig() =
+    default;
+AudioEncoderMultiChannelOpusConfig&
+AudioEncoderMultiChannelOpusConfig::operator=(
+    const AudioEncoderMultiChannelOpusConfig&) = default;
+
+bool AudioEncoderMultiChannelOpusConfig::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (num_channels >= 255) {
+    return false;
+  }
+  if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps)
+    return false;
+  if (complexity < 0 || complexity > 10)
+    return false;
+
+  // Check the lengths:
+  if (num_streams < 0 || coupled_streams < 0) {
+    return false;
+  }
+  if (num_streams < coupled_streams) {
+    return false;
+  }
+  if (channel_mapping.size() != static_cast<size_t>(num_channels)) {
+    return false;
+  }
+
+  // Every mono stream codes one channel, every coupled stream codes two. This
+  // is the total coded channel count:
+  const int max_coded_channel = num_streams + coupled_streams;
+  for (const auto& x : channel_mapping) {
+    // Coded channels >= max_coded_channel don't exist. Except for 255, which
+    // tells Opus to ignore input channel x.
+    if (x >= max_coded_channel && x != 255) {
+      return false;
+    }
+  }
+
+  // Inverse mapping.
+  constexpr int kNotSet = -1;
+  std::vector<int> coded_channels_to_input_channels(max_coded_channel, kNotSet);
+  for (size_t i = 0; i < num_channels; ++i) {
+    if (channel_mapping[i] == 255) {
+      continue;
+    }
+
+    // If it's not ignored, put it in the inverted mapping. But first check if
+    // we've told Opus to use another input channel for this coded channel:
+    const int coded_channel = channel_mapping[i];
+    if (coded_channels_to_input_channels[coded_channel] != kNotSet) {
+      // Coded channel `coded_channel` comes from both input channels
+      // `coded_channels_to_input_channels[coded_channel]` and `i`.
+      return false;
+    }
+
+    coded_channels_to_input_channels[coded_channel] = i;
+  }
+
+  // Check that we specified what input the encoder should use to produce
+  // every coded channel.
+  for (int i = 0; i < max_coded_channel; ++i) {
+    if (coded_channels_to_input_channels[i] == kNotSet) {
+      // Coded channel `i` has unspecified input channel.
+      return false;
+    }
+  }
+
+  if (num_channels > 255 || max_coded_channel >= 255) {
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct RTC_EXPORT AudioEncoderMultiChannelOpusConfig {
+  static constexpr int kDefaultFrameSizeMs = 20;
+
+  // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+  // bitrate should be in the range of 6000 to 510000, inclusive.
+  static constexpr int kMinBitrateBps = 6000;
+  static constexpr int kMaxBitrateBps = 510000;
+
+  AudioEncoderMultiChannelOpusConfig();
+  AudioEncoderMultiChannelOpusConfig(const AudioEncoderMultiChannelOpusConfig&);
+  ~AudioEncoderMultiChannelOpusConfig();
+  AudioEncoderMultiChannelOpusConfig& operator=(
+      const AudioEncoderMultiChannelOpusConfig&);
+
+  int frame_size_ms;
+  size_t num_channels;
+  enum class ApplicationMode { kVoip, kAudio };
+  ApplicationMode application;
+  int bitrate_bps;
+  bool fec_enabled;
+  bool cbr_enabled;
+  bool dtx_enabled;
+  int max_playback_rate_hz;
+  std::vector<int> supported_frame_lengths_ms;
+
+  int complexity;
+
+  // Number of mono/stereo Opus streams.
+  int num_streams;
+
+  // Number of channel pairs coupled together, see RFC 7845 section
+  // 5.1.1. Has to be less than the number of streams
+  int coupled_streams;
+
+  // Channel mapping table, defines the mapping from encoded streams to input
+  // channels. See RFC 7845 section 5.1.1.
+  std::vector<unsigned char> channel_mapping;
+
+  bool IsOk() const;
+};
+
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
+    const SdpAudioFormat& format) {
+  return AudioEncoderOpusImpl::SdpToConfig(format);
+}
+
+void AudioEncoderOpus::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  AudioEncoderOpusImpl::AppendSupportedEncoders(specs);
+}
+
+AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder(
+    const AudioEncoderOpusConfig& config) {
+  return AudioEncoderOpusImpl::QueryAudioEncoder(config);
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
+    const AudioEncoderOpusConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus.h

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderOpus {
+  using Config = AudioEncoderOpusConfig;
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+constexpr int kDefaultComplexity = 5;
+#else
+constexpr int kDefaultComplexity = 9;
+#endif
+
+constexpr int kDefaultLowRateComplexity =
+    WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity;
+
+}  // namespace
+
+constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+constexpr int AudioEncoderOpusConfig::kMinBitrateBps;
+constexpr int AudioEncoderOpusConfig::kMaxBitrateBps;
+
+AudioEncoderOpusConfig::AudioEncoderOpusConfig()
+    : frame_size_ms(kDefaultFrameSizeMs),
+      sample_rate_hz(48000),
+      num_channels(1),
+      application(ApplicationMode::kVoip),
+      bitrate_bps(32000),
+      fec_enabled(false),
+      cbr_enabled(false),
+      max_playback_rate_hz(48000),
+      complexity(kDefaultComplexity),
+      low_rate_complexity(kDefaultLowRateComplexity),
+      complexity_threshold_bps(12500),
+      complexity_threshold_window_bps(1500),
+      dtx_enabled(false),
+      uplink_bandwidth_update_interval_ms(200),
+      payload_type(-1) {}
+AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) =
+    default;
+AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default;
+AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=(
+    const AudioEncoderOpusConfig&) = default;
+
+bool AudioEncoderOpusConfig::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported input sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels >= 255) {
+    return false;
+  }
+  if (!bitrate_bps)
+    return false;
+  if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps)
+    return false;
+  if (complexity < 0 || complexity > 10)
+    return false;
+  if (low_rate_complexity < 0 || low_rate_complexity > 10)
+    return false;
+  return true;
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h

@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct RTC_EXPORT AudioEncoderOpusConfig {
+  static constexpr int kDefaultFrameSizeMs = 20;
+
+  // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+  // bitrate should be in the range of 6000 to 510000, inclusive.
+  static constexpr int kMinBitrateBps = 6000;
+  static constexpr int kMaxBitrateBps = 510000;
+
+  AudioEncoderOpusConfig();
+  AudioEncoderOpusConfig(const AudioEncoderOpusConfig&);
+  ~AudioEncoderOpusConfig();
+  AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&);
+
+  bool IsOk() const;  // Checks if the values are currently OK.
+
+  int frame_size_ms;
+  int sample_rate_hz;
+  size_t num_channels;
+  enum class ApplicationMode { kVoip, kAudio };
+  ApplicationMode application;
+
+  // NOTE: This member must always be set.
+  // TODO(kwiberg): Turn it into just an int.
+  absl::optional<int> bitrate_bps;
+
+  bool fec_enabled;
+  bool cbr_enabled;
+  int max_playback_rate_hz;
+
+  // `complexity` is used when the bitrate goes above
+  // `complexity_threshold_bps` + `complexity_threshold_window_bps`;
+  // `low_rate_complexity` is used when the bitrate falls below
+  // `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the
+  // interval in the middle, we keep using the most recent of the two
+  // complexity settings.
+  int complexity;
+  int low_rate_complexity;
+  int complexity_threshold_bps;
+  int complexity_threshold_window_bps;
+
+  bool dtx_enabled;
+  std::vector<int> supported_frame_lengths_ms;
+  int uplink_bandwidth_update_interval_ms;
+
+  // NOTE: This member isn't necessary, and will soon go away. See
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  int payload_type;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus_audio_decoder_factory.cc

@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus_audio_decoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+
+namespace webrtc {
+
+namespace {
+
+// Modify an audio decoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+    return T::MakeAudioDecoder(config, codec_pair_id);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDecoderFactory> CreateOpusAudioDecoderFactory() {
+  return CreateAudioDecoderFactory<
+      AudioDecoderOpus, NotAdvertised<AudioDecoderMultiChannelOpus>>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus_audio_decoder_factory.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create only Opus audio decoders. Works like
+// CreateAudioDecoderFactory<AudioDecoderOpus>(), but is easier to use and is
+// not inline because it isn't a template.
+rtc::scoped_refptr<AudioDecoderFactory> CreateOpusAudioDecoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus_audio_encoder_factory.cc

@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus_audio_encoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+
+namespace webrtc {
+namespace {
+
+// Modify an audio encoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static AudioCodecInfo QueryAudioEncoder(const Config& config) {
+    return T::QueryAudioEncoder(config);
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr) {
+    return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
+                               field_trials);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioEncoderFactory> CreateOpusAudioEncoderFactory() {
+  return CreateAudioEncoderFactory<
+      AudioEncoderOpus, NotAdvertised<AudioEncoderMultiChannelOpus>>();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/opus_audio_encoder_factory.h

@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create only Opus audio encoders. Works like
+// CreateAudioEncoderFactory<AudioEncoderOpus>(), but is easier to use and is
+// not inline because it isn't a template.
+rtc::scoped_refptr<AudioEncoderFactory> CreateOpusAudioEncoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/audio_codecs/test/BUILD.gn

@@ -0,0 +1,39 @@
+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+if (rtc_include_tests) {
+  rtc_library("audio_codecs_api_unittests") {
+    testonly = true
+    sources = [
+      "audio_decoder_factory_template_unittest.cc",
+      "audio_encoder_factory_template_unittest.cc",
+    ]
+    deps = [
+      "..:audio_codecs_api",
+      "../../../test:audio_codec_mocks",
+      "../../../test:scoped_key_value_config",
+      "../../../test:test_support",
+      "../L16:audio_decoder_L16",
+      "../L16:audio_encoder_L16",
+      "../g711:audio_decoder_g711",
+      "../g711:audio_encoder_g711",
+      "../g722:audio_decoder_g722",
+      "../g722:audio_encoder_g722",
+      "../ilbc:audio_decoder_ilbc",
+      "../ilbc:audio_encoder_ilbc",
+      "../opus:audio_decoder_opus",
+      "../opus:audio_encoder_opus",
+    ]
+  }
+}

TMessagesProj/jni/voip/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc

@@ -0,0 +1,222 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+
+#include <memory>
+
+#include "api/audio_codecs/L16/audio_decoder_L16.h"
+#include "api/audio_codecs/g711/audio_decoder_g711.h"
+#include "api/audio_codecs/g722/audio_decoder_g722.h"
+#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+namespace {
+
+struct BogusParams {
+  static SdpAudioFormat AudioFormat() { return {"bogus", 8000, 1}; }
+  static AudioCodecInfo CodecInfo() { return {8000, 1, 12345}; }
+};
+
+struct ShamParams {
+  static SdpAudioFormat AudioFormat() {
+    return {"sham", 16000, 2, {{"param", "value"}}};
+  }
+  static AudioCodecInfo CodecInfo() { return {16000, 2, 23456}; }
+};
+
+template <typename Params>
+struct AudioDecoderFakeApi {
+  struct Config {
+    SdpAudioFormat audio_format;
+  };
+
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    if (Params::AudioFormat() == audio_format) {
+      Config config = {audio_format};
+      return config;
+    } else {
+      return absl::nullopt;
+    }
+  }
+
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    specs->push_back({Params::AudioFormat(), Params::CodecInfo()});
+  }
+
+  static AudioCodecInfo QueryAudioDecoder(const Config&) {
+    return Params::CodecInfo();
+  }
+
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config&,
+      absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
+    auto dec = std::make_unique<testing::StrictMock<MockAudioDecoder>>();
+    EXPECT_CALL(*dec, SampleRateHz())
+        .WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz));
+    EXPECT_CALL(*dec, Die());
+    return std::move(dec);
+  }
+};
+
+}  // namespace
+
+TEST(AudioDecoderFactoryTemplateTest, NoDecoderTypes) {
+  test::ScopedKeyValueConfig field_trials;
+  rtc::scoped_refptr<AudioDecoderFactory> factory(
+      rtc::make_ref_counted<
+          audio_decoder_factory_template_impl::AudioDecoderFactoryT<>>(
+          &field_trials));
+  EXPECT_THAT(factory->GetSupportedDecoders(), ::testing::IsEmpty());
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTemplateTest, OneDecoderType) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderFakeApi<BogusParams>>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(8000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, TwoDecoderTypes) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderFakeApi<BogusParams>,
+                                           AudioDecoderFakeApi<ShamParams>>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}},
+                  AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}},
+                                 {16000, 2, 23456}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
+  EXPECT_TRUE(
+      factory->IsSupportedDecoder({"sham", 16000, 2, {{"param", "value"}}}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec1 = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec1);
+  EXPECT_EQ(8000, dec1->SampleRateHz());
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"sham", 16000, 2}, absl::nullopt));
+  auto dec2 = factory->MakeAudioDecoder(
+      {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
+  ASSERT_NE(nullptr, dec2);
+  EXPECT_EQ(16000, dec2->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, G711) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderG711>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
+                  AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"G711", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"PCMU", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"pcma", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"pcmu", 16000, 1}, absl::nullopt));
+  auto dec1 = factory->MakeAudioDecoder({"pcmu", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec1);
+  EXPECT_EQ(8000, dec1->SampleRateHz());
+  auto dec2 = factory->MakeAudioDecoder({"PCMA", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec2);
+  EXPECT_EQ(8000, dec2->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, G722) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderG722>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"G722", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec1);
+  EXPECT_EQ(16000, dec1->SampleRateHz());
+  EXPECT_EQ(1u, dec1->Channels());
+  auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, dec2);
+  EXPECT_EQ(16000, dec2->SampleRateHz());
+  EXPECT_EQ(2u, dec2->Channels());
+  auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3}, absl::nullopt);
+  ASSERT_EQ(nullptr, dec3);
+}
+
+TEST(AudioDecoderFactoryTemplateTest, Ilbc) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderIlbc>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13300}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"ilbc", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 8000, 1}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"ilbc", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(8000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, L16) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderL16>();
+  EXPECT_THAT(
+      factory->GetSupportedDecoders(),
+      ::testing::ElementsAre(
+          AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}},
+          AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}},
+          AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}},
+          AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"L16", 48000, 1}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"L16", 96000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"L16", 8000, 0}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"L16", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(48000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, Opus) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderOpus>();
+  AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
+  opus_info.allow_comfort_noise = false;
+  opus_info.supports_network_adaption = true;
+  const SdpAudioFormat opus_format(
+      {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(AudioCodecSpec{opus_format, opus_info}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"opus", 48000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"opus", 48000, 2}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"opus", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(48000, dec->SampleRateHz());
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc

@@ -0,0 +1,224 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+
+#include <memory>
+
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+namespace {
+
+struct BogusParams {
+  static SdpAudioFormat AudioFormat() { return {"bogus", 8000, 1}; }
+  static AudioCodecInfo CodecInfo() { return {8000, 1, 12345}; }
+};
+
+struct ShamParams {
+  static SdpAudioFormat AudioFormat() {
+    return {"sham", 16000, 2, {{"param", "value"}}};
+  }
+  static AudioCodecInfo CodecInfo() { return {16000, 2, 23456}; }
+};
+
+template <typename Params>
+struct AudioEncoderFakeApi {
+  struct Config {
+    SdpAudioFormat audio_format;
+  };
+
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    if (Params::AudioFormat() == audio_format) {
+      Config config = {audio_format};
+      return config;
+    } else {
+      return absl::nullopt;
+    }
+  }
+
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    specs->push_back({Params::AudioFormat(), Params::CodecInfo()});
+  }
+
+  static AudioCodecInfo QueryAudioEncoder(const Config&) {
+    return Params::CodecInfo();
+  }
+
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config&,
+      int payload_type,
+      absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
+    auto enc = std::make_unique<testing::StrictMock<MockAudioEncoder>>();
+    EXPECT_CALL(*enc, SampleRateHz())
+        .WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz));
+    return std::move(enc);
+  }
+};
+
+}  // namespace
+
+TEST(AudioEncoderFactoryTemplateTest, NoEncoderTypes) {
+  test::ScopedKeyValueConfig field_trials;
+  rtc::scoped_refptr<AudioEncoderFactory> factory(
+      rtc::make_ref_counted<
+          audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>(
+          &field_trials));
+  EXPECT_THAT(factory->GetSupportedEncoders(), ::testing::IsEmpty());
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+}
+
+TEST(AudioEncoderFactoryTemplateTest, OneEncoderType) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderFakeApi<BogusParams>>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
+            factory->QueryAudioEncoder({"bogus", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(8000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, TwoEncoderTypes) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderFakeApi<BogusParams>,
+                                           AudioEncoderFakeApi<ShamParams>>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}},
+                  AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}},
+                                 {16000, 2, 23456}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
+            factory->QueryAudioEncoder({"bogus", 8000, 1}));
+  EXPECT_EQ(
+      AudioCodecInfo(16000, 2, 23456),
+      factory->QueryAudioEncoder({"sham", 16000, 2, {{"param", "value"}}}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc1 = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc1);
+  EXPECT_EQ(8000, enc1->SampleRateHz());
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"sham", 16000, 2}, absl::nullopt));
+  auto enc2 = factory->MakeAudioEncoder(
+      17, {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
+  ASSERT_NE(nullptr, enc2);
+  EXPECT_EQ(16000, enc2->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, G711) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderG711>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
+                  AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 64000),
+            factory->QueryAudioEncoder({"PCMA", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"PCMU", 16000, 1}, absl::nullopt));
+  auto enc1 = factory->MakeAudioEncoder(17, {"PCMU", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc1);
+  EXPECT_EQ(8000, enc1->SampleRateHz());
+  auto enc2 = factory->MakeAudioEncoder(17, {"PCMA", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc2);
+  EXPECT_EQ(8000, enc2->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, G722) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderG722>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(16000, 1, 64000),
+            factory->QueryAudioEncoder({"G722", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(16000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, Ilbc) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderIlbc>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13333}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 13333),
+            factory->QueryAudioEncoder({"ilbc", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 8000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"ilbc", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(8000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, L16) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderL16>();
+  EXPECT_THAT(
+      factory->GetSupportedEncoders(),
+      ::testing::ElementsAre(
+          AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}},
+          AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}},
+          AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}},
+          AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
+  EXPECT_EQ(AudioCodecInfo(48000, 1, 48000 * 16),
+            factory->QueryAudioEncoder({"L16", 48000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"L16", 8000, 0}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(48000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, Opus) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
+  AudioCodecInfo info = {48000, 1, 32000, 6000, 510000};
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = true;
+  EXPECT_THAT(
+      factory->GetSupportedEncoders(),
+      ::testing::ElementsAre(AudioCodecSpec{
+          {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
+          info}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(
+      info,
+      factory->QueryAudioEncoder(
+          {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(48000, enc->SampleRateHz());
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/audio_options.cc

@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_options.h"
+
+#include "api/array_view.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace cricket {
+namespace {
+
+template <class T>
+void ToStringIfSet(rtc::SimpleStringBuilder* result,
+                   const char* key,
+                   const absl::optional<T>& val) {
+  if (val) {
+    (*result) << key << ": " << *val << ", ";
+  }
+}
+
+template <typename T>
+void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
+  if (o) {
+    *s = o;
+  }
+}
+
+}  // namespace
+
+AudioOptions::AudioOptions() = default;
+AudioOptions::~AudioOptions() = default;
+
+void AudioOptions::SetAll(const AudioOptions& change) {
+  SetFrom(&echo_cancellation, change.echo_cancellation);
+#if defined(WEBRTC_IOS)
+  SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
+#endif
+  SetFrom(&auto_gain_control, change.auto_gain_control);
+  SetFrom(&noise_suppression, change.noise_suppression);
+  SetFrom(&highpass_filter, change.highpass_filter);
+  SetFrom(&stereo_swapping, change.stereo_swapping);
+  SetFrom(&audio_jitter_buffer_max_packets,
+          change.audio_jitter_buffer_max_packets);
+  SetFrom(&audio_jitter_buffer_fast_accelerate,
+          change.audio_jitter_buffer_fast_accelerate);
+  SetFrom(&audio_jitter_buffer_min_delay_ms,
+          change.audio_jitter_buffer_min_delay_ms);
+  SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
+  SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
+  SetFrom(&init_recording_on_send, change.init_recording_on_send);
+}
+
+bool AudioOptions::operator==(const AudioOptions& o) const {
+  return echo_cancellation == o.echo_cancellation &&
+#if defined(WEBRTC_IOS)
+         ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
+#endif
+         auto_gain_control == o.auto_gain_control &&
+         noise_suppression == o.noise_suppression &&
+         highpass_filter == o.highpass_filter &&
+         stereo_swapping == o.stereo_swapping &&
+         audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
+         audio_jitter_buffer_fast_accelerate ==
+             o.audio_jitter_buffer_fast_accelerate &&
+         audio_jitter_buffer_min_delay_ms ==
+             o.audio_jitter_buffer_min_delay_ms &&
+         audio_network_adaptor == o.audio_network_adaptor &&
+         audio_network_adaptor_config == o.audio_network_adaptor_config &&
+         init_recording_on_send == o.init_recording_on_send;
+}
+
+std::string AudioOptions::ToString() const {
+  char buffer[1024];
+  rtc::SimpleStringBuilder result(buffer);
+  result << "AudioOptions {";
+  ToStringIfSet(&result, "aec", echo_cancellation);
+#if defined(WEBRTC_IOS)
+  ToStringIfSet(&result, "ios_force_software_aec_HACK",
+                ios_force_software_aec_HACK);
+#endif
+  ToStringIfSet(&result, "agc", auto_gain_control);
+  ToStringIfSet(&result, "ns", noise_suppression);
+  ToStringIfSet(&result, "hf", highpass_filter);
+  ToStringIfSet(&result, "swap", stereo_swapping);
+  ToStringIfSet(&result, "audio_jitter_buffer_max_packets",
+                audio_jitter_buffer_max_packets);
+  ToStringIfSet(&result, "audio_jitter_buffer_fast_accelerate",
+                audio_jitter_buffer_fast_accelerate);
+  ToStringIfSet(&result, "audio_jitter_buffer_min_delay_ms",
+                audio_jitter_buffer_min_delay_ms);
+  ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
+  ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);
+  result << "}";
+  return result.str();
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/api/audio_options.h

@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_AUDIO_OPTIONS_H_
+#define API_AUDIO_OPTIONS_H_
+
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace cricket {
+
+// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
+// Used to be flags, but that makes it hard to selectively apply options.
+// We are moving all of the setting of options to structs like this,
+// but some things currently still use flags.
+struct RTC_EXPORT AudioOptions {
+  AudioOptions();
+  ~AudioOptions();
+  void SetAll(const AudioOptions& change);
+
+  bool operator==(const AudioOptions& o) const;
+  bool operator!=(const AudioOptions& o) const { return !(*this == o); }
+
+  std::string ToString() const;
+
+  // Audio processing that attempts to filter away the output signal from
+  // later inbound pickup.
+  absl::optional<bool> echo_cancellation;
+#if defined(WEBRTC_IOS)
+  // Forces software echo cancellation on iOS. This is a temporary workaround
+  // (until Apple fixes the bug) for a device with non-functioning AEC. May
+  // improve performance on that particular device, but will cause unpredictable
+  // behavior in all other cases. See http://bugs.webrtc.org/8682.
+  absl::optional<bool> ios_force_software_aec_HACK;
+#endif
+  // Audio processing to adjust the sensitivity of the local mic dynamically.
+  absl::optional<bool> auto_gain_control;
+  // Audio processing to filter out background noise.
+  absl::optional<bool> noise_suppression;
+  // Audio processing to remove background noise of lower frequencies.
+  absl::optional<bool> highpass_filter;
+  // Audio processing to swap the left and right channels.
+  absl::optional<bool> stereo_swapping;
+  // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
+  absl::optional<int> audio_jitter_buffer_max_packets;
+  // Audio receiver jitter buffer (NetEq) fast accelerate mode.
+  absl::optional<bool> audio_jitter_buffer_fast_accelerate;
+  // Audio receiver jitter buffer (NetEq) minimum target delay in milliseconds.
+  absl::optional<int> audio_jitter_buffer_min_delay_ms;
+  // Enable audio network adaptor.
+  // TODO(webrtc:11717): Remove this API in favor of adaptivePtime in
+  // RtpEncodingParameters.
+  absl::optional<bool> audio_network_adaptor;
+  // Config string for audio network adaptor.
+  absl::optional<std::string> audio_network_adaptor_config;
+  // Pre-initialize the ADM for recording when starting to send. Default to
+  // true.
+  // TODO(webrtc:13566): Remove this option. See issue for details.
+  absl::optional<bool> init_recording_on_send;
+};
+
+}  // namespace cricket
+
+#endif  // API_AUDIO_OPTIONS_H_

TMessagesProj/jni/voip/webrtc/api/call/audio_sink.h

@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_CALL_AUDIO_SINK_H_
+#define API_CALL_AUDIO_SINK_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// Represents a simple push audio sink.
+class AudioSinkInterface {
+ public:
+  virtual ~AudioSinkInterface() {}
+
+  struct Data {
+    Data(const int16_t* data,
+         size_t samples_per_channel,
+         int sample_rate,
+         size_t channels,
+         uint32_t timestamp)
+        : data(data),
+          samples_per_channel(samples_per_channel),
+          sample_rate(sample_rate),
+          channels(channels),
+          timestamp(timestamp) {}
+
+    const int16_t* data;         // The actual 16bit audio data.
+    size_t samples_per_channel;  // Number of frames in the buffer.
+    int sample_rate;             // Sample rate in Hz.
+    size_t channels;             // Number of channels in the audio data.
+    uint32_t timestamp;          // The RTP timestamp of the first sample.
+  };
+
+  virtual void OnData(const Data& audio) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_CALL_AUDIO_SINK_H_

TMessagesProj/jni/voip/webrtc/api/call/bitrate_allocation.h

@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef API_CALL_BITRATE_ALLOCATION_H_
+#define API_CALL_BITRATE_ALLOCATION_H_
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+// BitrateAllocationUpdate provides information to allocated streams about their
+// bitrate allocation. It originates from the BitrateAllocater class and is
+// propagated from there.
+struct BitrateAllocationUpdate {
+  // The allocated target bitrate. Media streams should produce this amount of
+  // data. (Note that this may include packet overhead depending on
+  // configuration.)
+  DataRate target_bitrate = DataRate::Zero();
+  // The allocated part of the estimated link capacity. This is more stable than
+  // the target as it is based on the underlying link capacity estimate. This
+  // should be used to change encoder configuration when the cost of change is
+  // high.
+  DataRate stable_target_bitrate = DataRate::Zero();
+  // Predicted packet loss ratio.
+  double packet_loss_ratio = 0;
+  // Predicted round trip time.
+  TimeDelta round_trip_time = TimeDelta::PlusInfinity();
+  // `bwe_period` is deprecated, use `stable_target_bitrate` allocation instead.
+  TimeDelta bwe_period = TimeDelta::PlusInfinity();
+  // Congestion window pushback bitrate reduction fraction. Used in
+  // VideoStreamEncoder to reduce the bitrate by the given fraction
+  // by dropping frames.
+  double cwnd_reduce_ratio = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_CALL_BITRATE_ALLOCATION_H_

TMessagesProj/jni/voip/webrtc/api/call/transport.cc

@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/call/transport.h"
+
+#include <cstdint>
+
+namespace webrtc {
+
+PacketOptions::PacketOptions() = default;
+
+PacketOptions::PacketOptions(const PacketOptions&) = default;
+
+PacketOptions::~PacketOptions() = default;
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/call/transport.h

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_CALL_TRANSPORT_H_
+#define API_CALL_TRANSPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// TODO(holmer): Look into unifying this with the PacketOptions in
+// asyncpacketsocket.h.
+struct PacketOptions {
+  PacketOptions();
+  PacketOptions(const PacketOptions&);
+  ~PacketOptions();
+
+  // A 16 bits positive id. Negative ids are invalid and should be interpreted
+  // as packet_id not being set.
+  int packet_id = -1;
+  // Additional data bound to the RTP packet for use in application code,
+  // outside of WebRTC.
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
+  // Whether this is a retransmission of an earlier packet.
+  bool is_retransmit = false;
+  bool included_in_feedback = false;
+  bool included_in_allocation = false;
+  // Whether this packet can be part of a packet batch at lower levels.
+  bool batchable = false;
+  // Whether this packet is the last of a batch.
+  bool last_packet_in_batch = false;
+};
+
+class Transport {
+ public:
+  virtual bool SendRtp(rtc::ArrayView<const uint8_t> packet,
+                       const PacketOptions& options) = 0;
+  virtual bool SendRtcp(rtc::ArrayView<const uint8_t> packet) = 0;
+
+ protected:
+  virtual ~Transport() {}
+};
+
+}  // namespace webrtc
+
+#endif  // API_CALL_TRANSPORT_H_

TMessagesProj/jni/voip/webrtc/api/candidate.cc

@@ -0,0 +1,219 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/candidate.h"
+
+#include "absl/base/attributes.h"
+#include "rtc_base/helpers.h"
+#include "rtc_base/ip_address.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace cricket {
+
+ABSL_CONST_INIT const absl::string_view LOCAL_PORT_TYPE = "local";
+ABSL_CONST_INIT const absl::string_view STUN_PORT_TYPE = "stun";
+ABSL_CONST_INIT const absl::string_view PRFLX_PORT_TYPE = "prflx";
+ABSL_CONST_INIT const absl::string_view RELAY_PORT_TYPE = "relay";
+
+Candidate::Candidate()
+    : id_(rtc::CreateRandomString(8)),
+      component_(0),
+      priority_(0),
+      type_(LOCAL_PORT_TYPE),
+      network_type_(rtc::ADAPTER_TYPE_UNKNOWN),
+      underlying_type_for_vpn_(rtc::ADAPTER_TYPE_UNKNOWN),
+      generation_(0),
+      network_id_(0),
+      network_cost_(0) {}
+
+Candidate::Candidate(int component,
+                     absl::string_view protocol,
+                     const rtc::SocketAddress& address,
+                     uint32_t priority,
+                     absl::string_view username,
+                     absl::string_view password,
+                     absl::string_view type,
+                     uint32_t generation,
+                     absl::string_view foundation,
+                     uint16_t network_id,
+                     uint16_t network_cost)
+    : id_(rtc::CreateRandomString(8)),
+      component_(component),
+      protocol_(protocol),
+      address_(address),
+      priority_(priority),
+      username_(username),
+      password_(password),
+      type_(type),
+      network_type_(rtc::ADAPTER_TYPE_UNKNOWN),
+      underlying_type_for_vpn_(rtc::ADAPTER_TYPE_UNKNOWN),
+      generation_(generation),
+      foundation_(foundation),
+      network_id_(network_id),
+      network_cost_(network_cost) {}
+
+Candidate::Candidate(const Candidate&) = default;
+
+Candidate::~Candidate() = default;
+
+void Candidate::generate_id() {
+  id_ = rtc::CreateRandomString(8);
+}
+
+bool Candidate::is_local() const {
+  return type_ == LOCAL_PORT_TYPE;
+}
+bool Candidate::is_stun() const {
+  return type_ == STUN_PORT_TYPE;
+}
+bool Candidate::is_prflx() const {
+  return type_ == PRFLX_PORT_TYPE;
+}
+bool Candidate::is_relay() const {
+  return type_ == RELAY_PORT_TYPE;
+}
+
+absl::string_view Candidate::type_name() const {
+  // The LOCAL_PORT_TYPE and STUN_PORT_TYPE constants are not the standard type
+  // names, so check for those specifically. For other types, `type_` will have
+  // the correct name.
+  if (is_local())
+    return "host";
+  if (is_stun())
+    return "srflx";
+  return type_;
+}
+
+bool Candidate::IsEquivalent(const Candidate& c) const {
+  // We ignore the network name, since that is just debug information, and
+  // the priority and the network cost, since they should be the same if the
+  // rest are.
+  return (component_ == c.component_) && (protocol_ == c.protocol_) &&
+         (address_ == c.address_) && (username_ == c.username_) &&
+         (password_ == c.password_) && (type_ == c.type_) &&
+         (generation_ == c.generation_) && (foundation_ == c.foundation_) &&
+         (related_address_ == c.related_address_) &&
+         (network_id_ == c.network_id_);
+}
+
+bool Candidate::MatchesForRemoval(const Candidate& c) const {
+  return component_ == c.component_ && protocol_ == c.protocol_ &&
+         address_ == c.address_;
+}
+
+std::string Candidate::ToStringInternal(bool sensitive) const {
+  rtc::StringBuilder ost;
+  std::string address =
+      sensitive ? address_.ToSensitiveString() : address_.ToString();
+  std::string related_address = sensitive ? related_address_.ToSensitiveString()
+                                          : related_address_.ToString();
+  ost << "Cand[" << transport_name_ << ":" << foundation_ << ":" << component_
+      << ":" << protocol_ << ":" << priority_ << ":" << address << ":"
+      << type_name() << ":" << related_address << ":" << username_ << ":"
+      << password_ << ":" << network_id_ << ":" << network_cost_ << ":"
+      << generation_ << "]";
+  return ost.Release();
+}
+
+uint32_t Candidate::GetPriority(uint32_t type_preference,
+                                int network_adapter_preference,
+                                int relay_preference,
+                                bool adjust_local_preference) const {
+  // RFC 5245 - 4.1.2.1.
+  // priority = (2^24)*(type preference) +
+  //            (2^8)*(local preference) +
+  //            (2^0)*(256 - component ID)
+
+  // `local_preference` length is 2 bytes, 0-65535 inclusive.
+  // In our implemenation we will partion local_preference into
+  //              0                 1
+  //       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+  //      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  //      |  NIC Pref     |    Addr Pref  |
+  //      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // NIC Type - Type of the network adapter e.g. 3G/Wifi/Wired.
+  // Addr Pref - Address preference value as per RFC 3484.
+  // local preference =  (NIC Type << 8 | Addr_Pref) + relay preference.
+  // The relay preference is based on the number of TURN servers, the
+  // first TURN server gets the highest preference.
+  int addr_pref = IPAddressPrecedence(address_.ipaddr());
+  int local_preference =
+      ((network_adapter_preference << 8) | addr_pref) + relay_preference;
+
+  // Ensure that the added relay preference will not result in a relay candidate
+  // whose STUN priority attribute has a higher priority than a server-reflexive
+  // candidate.
+  // The STUN priority attribute is calculated as
+  // (peer-reflexive type preference) << 24 | (priority & 0x00FFFFFF)
+  // as described in
+  // https://www.rfc-editor.org/rfc/rfc5245#section-7.1.2.1
+  // To satisfy that condition, add kMaxTurnServers to the local preference.
+  // This can not overflow the field width since the highest "NIC pref"
+  // assigned is kHighestNetworkPreference = 127
+  RTC_DCHECK_LT(local_preference + kMaxTurnServers, 0x10000);
+  if (adjust_local_preference && relay_protocol_.empty()) {
+    local_preference += kMaxTurnServers;
+  }
+
+  return (type_preference << 24) | (local_preference << 8) | (256 - component_);
+}
+
+bool Candidate::operator==(const Candidate& o) const {
+  return id_ == o.id_ && component_ == o.component_ &&
+         protocol_ == o.protocol_ && relay_protocol_ == o.relay_protocol_ &&
+         address_ == o.address_ && priority_ == o.priority_ &&
+         username_ == o.username_ && password_ == o.password_ &&
+         type_ == o.type_ && network_name_ == o.network_name_ &&
+         network_type_ == o.network_type_ && generation_ == o.generation_ &&
+         foundation_ == o.foundation_ &&
+         related_address_ == o.related_address_ && tcptype_ == o.tcptype_ &&
+         transport_name_ == o.transport_name_ && network_id_ == o.network_id_;
+}
+
+bool Candidate::operator!=(const Candidate& o) const {
+  return !(*this == o);
+}
+
+Candidate Candidate::ToSanitizedCopy(bool use_hostname_address,
+                                     bool filter_related_address) const {
+  Candidate copy(*this);
+  if (use_hostname_address) {
+    rtc::IPAddress ip;
+    if (address().hostname().empty()) {
+      // IP needs to be redacted, but no hostname available.
+      rtc::SocketAddress redacted_addr("redacted-ip.invalid", address().port());
+      copy.set_address(redacted_addr);
+    } else if (IPFromString(address().hostname(), &ip)) {
+      // The hostname is an IP literal, and needs to be redacted too.
+      rtc::SocketAddress redacted_addr("redacted-literal.invalid",
+                                       address().port());
+      copy.set_address(redacted_addr);
+    } else {
+      rtc::SocketAddress hostname_only_addr(address().hostname(),
+                                            address().port());
+      copy.set_address(hostname_only_addr);
+    }
+  }
+  if (filter_related_address) {
+    copy.set_related_address(
+        rtc::EmptySocketAddressWithFamily(copy.address().family()));
+  }
+  return copy;
+}
+
+void Candidate::Assign(std::string& s, absl::string_view view) {
+  // Assigning via a temporary object, like s = std::string(view), results in
+  // binary size bloat. To avoid that, extract pointer and size from the
+  // string view, and use std::string::assign method.
+  s.assign(view.data(), view.size());
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/api/candidate.h

@@ -0,0 +1,255 @@
+/*
+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_CANDIDATE_H_
+#define API_CANDIDATE_H_
+
+#include <limits.h>
+#include <stdint.h>
+
+#include <algorithm>
+#include <string>
+
+#include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/network_constants.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+enum class IceCandidateType : int { kHost, kSrflx, kPrflx, kRelay };
+}  // namespace webrtc
+
+namespace cricket {
+
+// TODO(tommi): These are temporarily here, moved from `port.h` and will
+// eventually be removed once we use enums instead of strings for these values.
+RTC_EXPORT extern const absl::string_view LOCAL_PORT_TYPE;
+RTC_EXPORT extern const absl::string_view STUN_PORT_TYPE;
+RTC_EXPORT extern const absl::string_view PRFLX_PORT_TYPE;
+RTC_EXPORT extern const absl::string_view RELAY_PORT_TYPE;
+
+// TURN servers are limited to 32 in accordance with
+// https://w3c.github.io/webrtc-pc/#dom-rtcconfiguration-iceservers
+static constexpr size_t kMaxTurnServers = 32;
+
+// Candidate for ICE based connection discovery.
+// TODO(phoglund): remove things in here that are not needed in the public API.
+
+class RTC_EXPORT Candidate {
+ public:
+  Candidate();
+  // TODO(pthatcher): Match the ordering and param list as per RFC 5245
+  // candidate-attribute syntax. http://tools.ietf.org/html/rfc5245#section-15.1
+  Candidate(int component,
+            absl::string_view protocol,
+            const rtc::SocketAddress& address,
+            uint32_t priority,
+            absl::string_view username,
+            absl::string_view password,
+            absl::string_view type ABSL_ATTRIBUTE_LIFETIME_BOUND,
+            uint32_t generation,
+            absl::string_view foundation,
+            uint16_t network_id = 0,
+            uint16_t network_cost = 0);
+  Candidate(const Candidate&);
+  ~Candidate();
+
+  // 8 character long randomized ID string for logging purposes.
+  const std::string& id() const { return id_; }
+  // Generates a new, 8 character long, id.
+  void generate_id();
+  // TODO(tommi): Callers should use generate_id(). Remove.
+  [[deprecated]] void set_id(absl::string_view id) { Assign(id_, id); }
+
+  int component() const { return component_; }
+  void set_component(int component) { component_ = component; }
+
+  const std::string& protocol() const { return protocol_; }
+  void set_protocol(absl::string_view protocol) { Assign(protocol_, protocol); }
+
+  // The protocol used to talk to relay.
+  const std::string& relay_protocol() const { return relay_protocol_; }
+  void set_relay_protocol(absl::string_view protocol) {
+    Assign(relay_protocol_, protocol);
+  }
+
+  const rtc::SocketAddress& address() const { return address_; }
+  void set_address(const rtc::SocketAddress& address) { address_ = address; }
+
+  uint32_t priority() const { return priority_; }
+  void set_priority(const uint32_t priority) { priority_ = priority; }
+
+  // TODO(honghaiz): Change to usernameFragment or ufrag.
+  const std::string& username() const { return username_; }
+  void set_username(absl::string_view username) { Assign(username_, username); }
+
+  const std::string& password() const { return password_; }
+  void set_password(absl::string_view password) { Assign(password_, password); }
+
+  const std::string& type() const { return type_; }
+
+  // Returns the name of the candidate type as specified in
+  // https://datatracker.ietf.org/doc/html/rfc5245#section-15.1
+  absl::string_view type_name() const;
+
+  // Setting the type requires a constant string (e.g.
+  // cricket::LOCAL_PORT_TYPE). The type should really be an enum rather than a
+  // string, but until we make that change the lifetime attribute helps us lock
+  // things down. See also the `Port` class.
+  void set_type(absl::string_view type ABSL_ATTRIBUTE_LIFETIME_BOUND) {
+    Assign(type_, type);
+  }
+
+  // Provide these simple checkers to abstract away dependency on the port types
+  // that are currently defined outside of Candidate. This will ease the change
+  // from the string type to an enum.
+  bool is_local() const;
+  bool is_stun() const;
+  bool is_prflx() const;
+  bool is_relay() const;
+
+  // Returns the type preference, a value between 0-126 inclusive, with 0 being
+  // the lowest preference value, as described in RFC 5245.
+  // https://datatracker.ietf.org/doc/html/rfc5245#section-4.1.2.1
+  int type_preference() const {
+    // From https://datatracker.ietf.org/doc/html/rfc5245#section-4.1.4 :
+    // It is RECOMMENDED that default candidates be chosen based on the
+    // likelihood of those candidates to work with the peer that is being
+    // contacted.
+    // I.e. it is recommended that relayed > reflexive > host.
+    if (is_local())
+      return 1;  // Host.
+    if (is_stun())
+      return 2;  // Reflexive.
+    if (is_relay())
+      return 3;  // Relayed.
+    return 0;    // Unknown, lowest preference.
+  }
+
+  const std::string& network_name() const { return network_name_; }
+  void set_network_name(absl::string_view network_name) {
+    Assign(network_name_, network_name);
+  }
+
+  rtc::AdapterType network_type() const { return network_type_; }
+  void set_network_type(rtc::AdapterType network_type) {
+    network_type_ = network_type;
+  }
+
+  rtc::AdapterType underlying_type_for_vpn() const {
+    return underlying_type_for_vpn_;
+  }
+  void set_underlying_type_for_vpn(rtc::AdapterType network_type) {
+    underlying_type_for_vpn_ = network_type;
+  }
+
+  // Candidates in a new generation replace those in the old generation.
+  uint32_t generation() const { return generation_; }
+  void set_generation(uint32_t generation) { generation_ = generation; }
+
+  // `network_cost` measures the cost/penalty of using this candidate. A network
+  // cost of 0 indicates this candidate can be used freely. A value of
+  // rtc::kNetworkCostMax indicates it should be used only as the last resort.
+  void set_network_cost(uint16_t network_cost) {
+    RTC_DCHECK_LE(network_cost, rtc::kNetworkCostMax);
+    network_cost_ = network_cost;
+  }
+  uint16_t network_cost() const { return network_cost_; }
+
+  // An ID assigned to the network hosting the candidate.
+  uint16_t network_id() const { return network_id_; }
+  void set_network_id(uint16_t network_id) { network_id_ = network_id; }
+
+  const std::string& foundation() const { return foundation_; }
+  void set_foundation(absl::string_view foundation) {
+    Assign(foundation_, foundation);
+  }
+
+  const rtc::SocketAddress& related_address() const { return related_address_; }
+  void set_related_address(const rtc::SocketAddress& related_address) {
+    related_address_ = related_address;
+  }
+  const std::string& tcptype() const { return tcptype_; }
+  void set_tcptype(absl::string_view tcptype) { Assign(tcptype_, tcptype); }
+
+  // The name of the transport channel of this candidate.
+  // TODO(phoglund): remove.
+  const std::string& transport_name() const { return transport_name_; }
+  void set_transport_name(absl::string_view transport_name) {
+    Assign(transport_name_, transport_name);
+  }
+
+  // The URL of the ICE server which this candidate is gathered from.
+  const std::string& url() const { return url_; }
+  void set_url(absl::string_view url) { Assign(url_, url); }
+
+  // Determines whether this candidate is equivalent to the given one.
+  bool IsEquivalent(const Candidate& c) const;
+
+  // Determines whether this candidate can be considered equivalent to the
+  // given one when looking for a matching candidate to remove.
+  bool MatchesForRemoval(const Candidate& c) const;
+
+  std::string ToString() const { return ToStringInternal(false); }
+
+  std::string ToSensitiveString() const { return ToStringInternal(true); }
+
+  uint32_t GetPriority(uint32_t type_preference,
+                       int network_adapter_preference,
+                       int relay_preference,
+                       bool adjust_local_preference) const;
+
+  bool operator==(const Candidate& o) const;
+  bool operator!=(const Candidate& o) const;
+
+  // Returns a sanitized copy configured by the given booleans. If
+  // `use_host_address` is true, the returned copy has its IP removed from
+  // `address()`, which leads `address()` to be a hostname address. If
+  // `filter_related_address`, the returned copy has its related address reset
+  // to the wildcard address (i.e. 0.0.0.0 for IPv4 and :: for IPv6). Note that
+  // setting both booleans to false returns an identical copy to the original
+  // candidate.
+  Candidate ToSanitizedCopy(bool use_hostname_address,
+                            bool filter_related_address) const;
+
+ private:
+  // TODO(bugs.webrtc.org/13220): With C++17, we get a std::string assignment
+  // operator accepting any object implicitly convertible to std::string_view,
+  // and then we don't need this workaround.
+  static void Assign(std::string& s, absl::string_view view);
+  std::string ToStringInternal(bool sensitive) const;
+
+  std::string id_;
+  int component_;
+  std::string protocol_;
+  std::string relay_protocol_;
+  rtc::SocketAddress address_;
+  uint32_t priority_;
+  std::string username_;
+  std::string password_;
+  std::string type_;
+  std::string network_name_;
+  rtc::AdapterType network_type_;
+  rtc::AdapterType underlying_type_for_vpn_;
+  uint32_t generation_;
+  std::string foundation_;
+  rtc::SocketAddress related_address_;
+  std::string tcptype_;
+  std::string transport_name_;
+  uint16_t network_id_;
+  uint16_t network_cost_;
+  std::string url_;
+};
+
+}  // namespace cricket
+
+#endif  // API_CANDIDATE_H_

TMessagesProj/jni/voip/webrtc/api/candidate_unittest.cc

@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2024 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/candidate.h"
+
+#include <string>
+
+#include "p2p/base/p2p_constants.h"
+#include "rtc_base/gunit.h"
+
+namespace cricket {
+
+TEST(CandidateTest, Id) {
+  Candidate c;
+  EXPECT_EQ(c.id().size(), 8u);
+  std::string current_id = c.id();
+  // Generate a new ID.
+  c.generate_id();
+  EXPECT_EQ(c.id().size(), 8u);
+  EXPECT_NE(current_id, c.id());
+}
+
+TEST(CandidateTest, Component) {
+  Candidate c;
+  EXPECT_EQ(c.component(), 0);
+  c.set_component(ICE_CANDIDATE_COMPONENT_DEFAULT);
+  EXPECT_EQ(c.component(), ICE_CANDIDATE_COMPONENT_DEFAULT);
+}
+
+TEST(CandidateTest, TypeName) {
+  Candidate c;
+  // The `type_name()` property defaults to "host".
+  EXPECT_EQ(c.type_name(), "host");
+  EXPECT_EQ(c.type(), LOCAL_PORT_TYPE);
+
+  c.set_type(STUN_PORT_TYPE);
+  EXPECT_EQ(c.type_name(), "srflx");
+  EXPECT_EQ(c.type(), STUN_PORT_TYPE);
+
+  c.set_type(PRFLX_PORT_TYPE);
+  EXPECT_EQ(c.type_name(), "prflx");
+  EXPECT_EQ(c.type(), PRFLX_PORT_TYPE);
+
+  c.set_type(RELAY_PORT_TYPE);
+  EXPECT_EQ(c.type_name(), "relay");
+  EXPECT_EQ(c.type(), RELAY_PORT_TYPE);
+}
+
+}  // namespace cricket

TMessagesProj/jni/voip/webrtc/api/create_peerconnection_factory.cc

@@ -0,0 +1,75 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/create_peerconnection_factory.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/enable_media.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/field_trial_based_config.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
+    rtc::Thread* network_thread,
+    rtc::Thread* worker_thread,
+    rtc::Thread* signaling_thread,
+    rtc::scoped_refptr<AudioDeviceModule> default_adm,
+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
+    std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
+    std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
+    rtc::scoped_refptr<AudioMixer> audio_mixer,
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    std::unique_ptr<AudioFrameProcessor> audio_frame_processor,
+    std::unique_ptr<FieldTrialsView> field_trials) {
+  if (!field_trials) {
+    field_trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
+  }
+
+  PeerConnectionFactoryDependencies dependencies;
+  dependencies.network_thread = network_thread;
+  dependencies.worker_thread = worker_thread;
+  dependencies.signaling_thread = signaling_thread;
+  dependencies.task_queue_factory =
+      CreateDefaultTaskQueueFactory(field_trials.get());
+  dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>();
+  dependencies.trials = std::move(field_trials);
+
+  if (network_thread) {
+    // TODO(bugs.webrtc.org/13145): Add an rtc::SocketFactory* argument.
+    dependencies.socket_factory = network_thread->socketserver();
+  }
+  dependencies.adm = std::move(default_adm);
+  dependencies.audio_encoder_factory = std::move(audio_encoder_factory);
+  dependencies.audio_decoder_factory = std::move(audio_decoder_factory);
+  dependencies.audio_frame_processor = std::move(audio_frame_processor);
+  if (audio_processing) {
+    dependencies.audio_processing = std::move(audio_processing);
+  } else {
+    dependencies.audio_processing = AudioProcessingBuilder().Create();
+  }
+  dependencies.audio_mixer = std::move(audio_mixer);
+  dependencies.video_encoder_factory = std::move(video_encoder_factory);
+  dependencies.video_decoder_factory = std::move(video_decoder_factory);
+  EnableMedia(dependencies);
+
+  return CreateModularPeerConnectionFactory(std::move(dependencies));
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/api/create_peerconnection_factory.h

@@ -0,0 +1,57 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_CREATE_PEERCONNECTION_FACTORY_H_
+#define API_CREATE_PEERCONNECTION_FACTORY_H_
+
+#include <memory>
+
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+
+namespace rtc {
+// TODO(bugs.webrtc.org/9987): Move rtc::Thread to api/ or expose a better
+// type. At the moment, rtc::Thread is not part of api/ so it cannot be
+// included in order to avoid to leak internal types.
+class Thread;
+}  // namespace rtc
+
+namespace webrtc {
+
+class AudioDeviceModule;
+class AudioFrameProcessor;
+class AudioProcessing;
+
+// Create a new instance of PeerConnectionFactoryInterface with optional video
+// codec factories. These video factories represents all video codecs, i.e. no
+// extra internal video codecs will be added.
+RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
+CreatePeerConnectionFactory(
+    rtc::Thread* network_thread,
+    rtc::Thread* worker_thread,
+    rtc::Thread* signaling_thread,
+    rtc::scoped_refptr<AudioDeviceModule> default_adm,
+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
+    std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
+    std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
+    rtc::scoped_refptr<AudioMixer> audio_mixer,
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    std::unique_ptr<AudioFrameProcessor> audio_frame_processor = nullptr,
+    std::unique_ptr<FieldTrialsView> field_trials = nullptr);
+
+}  // namespace webrtc
+
+#endif  // API_CREATE_PEERCONNECTION_FACTORY_H_

TMessagesProj/jni/voip/webrtc/api/crypto/BUILD.gn

@@ -0,0 +1,49 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+group("crypto") {
+  deps = [
+    ":frame_decryptor_interface",
+    ":frame_encryptor_interface",
+    ":options",
+  ]
+}
+
+rtc_library("options") {
+  visibility = [ "*" ]
+  sources = [
+    "crypto_options.cc",
+    "crypto_options.h",
+  ]
+  deps = [
+    "../../rtc_base:ssl",
+    "../../rtc_base/system:rtc_export",
+  ]
+}
+
+rtc_source_set("frame_decryptor_interface") {
+  visibility = [ "*" ]
+  sources = [ "frame_decryptor_interface.h" ]
+  deps = [
+    "..:array_view",
+    "..:rtp_parameters",
+    "../../rtc_base:refcount",
+  ]
+}
+
+rtc_source_set("frame_encryptor_interface") {
+  visibility = [ "*" ]
+  sources = [ "frame_encryptor_interface.h" ]
+  deps = [
+    "..:array_view",
+    "..:rtp_parameters",
+    "../../rtc_base:refcount",
+  ]
+}