Repo created

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

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+# 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

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+gustaf@webrtc.org
+peah@webrtc.org

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

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+/*
+ *  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

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+/*
+ *  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

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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.
+ */
+
+#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

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+/*
+ *  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