Repo created

TMessagesProj/jni/voip/webrtc/system_wrappers/source/clock.cc

@@ -0,0 +1,102 @@
+/*
+ *  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.
+ */
+
+#include "system_wrappers/include/clock.h"
+
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace {
+
+int64_t NtpOffsetUsCalledOnce() {
+  constexpr int64_t kNtpJan1970Sec = 2208988800;
+  int64_t clock_time = rtc::TimeMicros();
+  int64_t utc_time = rtc::TimeUTCMicros();
+  return utc_time - clock_time + kNtpJan1970Sec * rtc::kNumMicrosecsPerSec;
+}
+
+NtpTime TimeMicrosToNtp(int64_t time_us) {
+  static int64_t ntp_offset_us = NtpOffsetUsCalledOnce();
+
+  int64_t time_ntp_us = time_us + ntp_offset_us;
+  RTC_DCHECK_GE(time_ntp_us, 0);  // Time before year 1900 is unsupported.
+
+  // Convert seconds to uint32 through uint64 for a well-defined cast.
+  // A wrap around, which will happen in 2036, is expected for NTP time.
+  uint32_t ntp_seconds =
+      static_cast<uint64_t>(time_ntp_us / rtc::kNumMicrosecsPerSec);
+
+  // Scale fractions of the second to NTP resolution.
+  constexpr int64_t kNtpFractionsInSecond = 1LL << 32;
+  int64_t us_fractions = time_ntp_us % rtc::kNumMicrosecsPerSec;
+  uint32_t ntp_fractions =
+      us_fractions * kNtpFractionsInSecond / rtc::kNumMicrosecsPerSec;
+
+  return NtpTime(ntp_seconds, ntp_fractions);
+}
+
+}  // namespace
+
+class RealTimeClock : public Clock {
+ public:
+  RealTimeClock() = default;
+
+  Timestamp CurrentTime() override {
+    return Timestamp::Micros(rtc::TimeMicros());
+  }
+
+  NtpTime ConvertTimestampToNtpTime(Timestamp timestamp) override {
+    return TimeMicrosToNtp(timestamp.us());
+  }
+};
+
+Clock* Clock::GetRealTimeClock() {
+  static Clock* const clock = new RealTimeClock();
+  return clock;
+}
+
+SimulatedClock::SimulatedClock(int64_t initial_time_us)
+    : time_us_(initial_time_us) {}
+
+SimulatedClock::SimulatedClock(Timestamp initial_time)
+    : SimulatedClock(initial_time.us()) {}
+
+SimulatedClock::~SimulatedClock() {}
+
+Timestamp SimulatedClock::CurrentTime() {
+  return Timestamp::Micros(time_us_.load(std::memory_order_relaxed));
+}
+
+NtpTime SimulatedClock::ConvertTimestampToNtpTime(Timestamp timestamp) {
+  int64_t now_us = timestamp.us();
+  uint32_t seconds = (now_us / 1'000'000) + kNtpJan1970;
+  uint32_t fractions = static_cast<uint32_t>(
+      (now_us % 1'000'000) * kMagicNtpFractionalUnit / 1'000'000);
+  return NtpTime(seconds, fractions);
+}
+
+void SimulatedClock::AdvanceTimeMilliseconds(int64_t milliseconds) {
+  AdvanceTime(TimeDelta::Millis(milliseconds));
+}
+
+void SimulatedClock::AdvanceTimeMicroseconds(int64_t microseconds) {
+  AdvanceTime(TimeDelta::Micros(microseconds));
+}
+
+// TODO(bugs.webrtc.org(12102): It's desirable to let a single thread own
+// advancement of the clock. We could then replace this read-modify-write
+// operation with just a thread checker. But currently, that breaks a couple of
+// tests, in particular, RepeatingTaskTest.ClockIntegration and
+// CallStatsTest.LastProcessedRtt.
+void SimulatedClock::AdvanceTime(TimeDelta delta) {
+  time_us_.fetch_add(delta.us(), std::memory_order_relaxed);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/cpu_features.cc

@@ -0,0 +1,119 @@
+/*
+ *  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.
+ */
+
+// Parts of this file derived from Chromium's base/cpu.cc.
+
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "system_wrappers/include/field_trial.h"
+
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(_MSC_VER)
+#include <intrin.h>
+#endif
+
+namespace webrtc {
+
+// No CPU feature is available => straight C path.
+int GetCPUInfoNoASM(CPUFeature feature) {
+  (void)feature;
+  return 0;
+}
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+
+#if defined(WEBRTC_ENABLE_AVX2)
+// xgetbv returns the value of an Intel Extended Control Register (XCR).
+// Currently only XCR0 is defined by Intel so `xcr` should always be zero.
+static uint64_t xgetbv(uint32_t xcr) {
+#if defined(_MSC_VER)
+  return _xgetbv(xcr);
+#else
+  uint32_t eax, edx;
+
+  __asm__ volatile("xgetbv" : "=a"(eax), "=d"(edx) : "c"(xcr));
+  return (static_cast<uint64_t>(edx) << 32) | eax;
+#endif  // _MSC_VER
+}
+#endif  // WEBRTC_ENABLE_AVX2
+
+#ifndef _MSC_VER
+// Intrinsic for "cpuid".
+#if defined(__pic__) && defined(__i386__)
+static inline void __cpuid(int cpu_info[4], int info_type) {
+  __asm__ volatile(
+      "mov %%ebx, %%edi\n"
+      "cpuid\n"
+      "xchg %%edi, %%ebx\n"
+      : "=a"(cpu_info[0]), "=D"(cpu_info[1]), "=c"(cpu_info[2]),
+        "=d"(cpu_info[3])
+      : "a"(info_type));
+}
+#else
+static inline void __cpuid(int cpu_info[4], int info_type) {
+  __asm__ volatile("cpuid\n"
+                   : "=a"(cpu_info[0]), "=b"(cpu_info[1]), "=c"(cpu_info[2]),
+                     "=d"(cpu_info[3])
+                   : "a"(info_type), "c"(0));
+}
+#endif
+#endif  // _MSC_VER
+#endif  // WEBRTC_ARCH_X86_FAMILY
+
+#if defined(WEBRTC_ARCH_X86_FAMILY)
+// Actual feature detection for x86.
+int GetCPUInfo(CPUFeature feature) {
+  int cpu_info[4];
+  __cpuid(cpu_info, 1);
+  if (feature == kSSE2) {
+    return 0 != (cpu_info[3] & 0x04000000);
+  }
+  if (feature == kSSE3) {
+    return 0 != (cpu_info[2] & 0x00000001);
+  }
+#if defined(WEBRTC_ENABLE_AVX2)
+  if (feature == kAVX2 &&
+      !webrtc::field_trial::IsEnabled("WebRTC-Avx2SupportKillSwitch")) {
+    int cpu_info7[4];
+    __cpuid(cpu_info7, 0);
+    int num_ids = cpu_info7[0];
+    if (num_ids < 7) {
+      return 0;
+    }
+    // Interpret CPU feature information.
+    __cpuid(cpu_info7, 7);
+
+    // AVX instructions can be used when
+    //     a) AVX are supported by the CPU,
+    //     b) XSAVE is supported by the CPU,
+    //     c) XSAVE is enabled by the kernel.
+    // Compiling with MSVC and /arch:AVX2 surprisingly generates BMI2
+    // instructions (see crbug.com/1315519).
+    return (cpu_info[2] & 0x10000000) != 0 /* AVX */ &&
+           (cpu_info[2] & 0x04000000) != 0 /* XSAVE */ &&
+           (cpu_info[2] & 0x08000000) != 0 /* OSXSAVE */ &&
+           (xgetbv(0) & 0x00000006) == 6 /* XSAVE enabled by kernel */ &&
+           (cpu_info7[1] & 0x00000020) != 0 /* AVX2 */ &&
+           (cpu_info7[1] & 0x00000100) != 0 /* BMI2 */;
+  }
+#endif  // WEBRTC_ENABLE_AVX2
+  if (feature == kFMA3) {
+    return 0 != (cpu_info[2] & 0x00001000);
+  }
+  return 0;
+}
+#else
+// Default to straight C for other platforms.
+int GetCPUInfo(CPUFeature feature) {
+  (void)feature;
+  return 0;
+}
+#endif
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/cpu_features_android.cc

@@ -0,0 +1,19 @@
+/*
+ *  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 <cpu-features.h>
+
+namespace webrtc {
+
+uint64_t GetCPUFeaturesARM(void) {
+  return android_getCpuFeatures();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/cpu_features_linux.cc

@@ -0,0 +1,97 @@
+/*
+ *  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 <features.h>
+#include <stdlib.h>
+#include <string.h>
+
+#ifdef __GLIBC_PREREQ
+#define WEBRTC_GLIBC_PREREQ(a, b) __GLIBC_PREREQ(a, b)
+#else
+#define WEBRTC_GLIBC_PREREQ(a, b) 0
+#endif
+
+#if WEBRTC_GLIBC_PREREQ(2, 16)
+#include <sys/auxv.h>
+#else
+#include <errno.h>
+#include <fcntl.h>
+#include <link.h>
+#include <unistd.h>
+#endif
+
+#include "rtc_base/system/arch.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+
+#if defined(WEBRTC_ARCH_ARM_FAMILY)
+#include <asm/hwcap.h>
+
+namespace webrtc {
+
+uint64_t GetCPUFeaturesARM(void) {
+  uint64_t result = 0;
+  int architecture = 0;
+  uint64_t hwcap = 0;
+  const char* platform = NULL;
+#if WEBRTC_GLIBC_PREREQ(2, 16)
+  hwcap = getauxval(AT_HWCAP);
+  platform = (const char*)getauxval(AT_PLATFORM);
+#else
+  ElfW(auxv_t) auxv;
+  int fd = open("/proc/self/auxv", O_RDONLY);
+  if (fd >= 0) {
+    while (hwcap == 0 || platform == NULL) {
+      if (read(fd, &auxv, sizeof(auxv)) < (ssize_t)sizeof(auxv)) {
+        if (errno == EINTR)
+          continue;
+        break;
+      }
+      switch (auxv.a_type) {
+        case AT_HWCAP:
+          hwcap = auxv.a_un.a_val;
+          break;
+        case AT_PLATFORM:
+          platform = (const char*)auxv.a_un.a_val;
+          break;
+      }
+    }
+    close(fd);
+  }
+#endif  // WEBRTC_GLIBC_PREREQ(2, 16)
+#if defined(__aarch64__)
+  (void)platform;
+  architecture = 8;
+  if ((hwcap & HWCAP_FP) != 0)
+    result |= kCPUFeatureVFPv3;
+  if ((hwcap & HWCAP_ASIMD) != 0)
+    result |= kCPUFeatureNEON;
+#else
+  if (platform != NULL) {
+    /* expect a string in the form "v6l" or "v7l", etc.
+     */
+    if (platform[0] == 'v' && '0' <= platform[1] && platform[1] <= '9' &&
+        (platform[2] == 'l' || platform[2] == 'b')) {
+      architecture = platform[1] - '0';
+    }
+  }
+  if ((hwcap & HWCAP_VFPv3) != 0)
+    result |= kCPUFeatureVFPv3;
+  if ((hwcap & HWCAP_NEON) != 0)
+    result |= kCPUFeatureNEON;
+#endif
+  if (architecture >= 7)
+    result |= kCPUFeatureARMv7;
+  if (architecture >= 6)
+    result |= kCPUFeatureLDREXSTREX;
+  return result;
+}
+
+}  // namespace webrtc
+#endif  // WEBRTC_ARCH_ARM_FAMILY

TMessagesProj/jni/voip/webrtc/system_wrappers/source/cpu_info.cc

@@ -0,0 +1,72 @@
+/*
+ *  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.
+ */
+
+#include "system_wrappers/include/cpu_info.h"
+
+#if defined(WEBRTC_WIN)
+#include <windows.h>
+#elif defined(WEBRTC_LINUX)
+#include <unistd.h>
+#elif defined(WEBRTC_MAC)
+#include <sys/sysctl.h>
+#elif defined(WEBRTC_FUCHSIA)
+#include <zircon/syscalls.h>
+#endif
+
+#include "rtc_base/logging.h"
+
+namespace internal {
+static int DetectNumberOfCores() {
+  int number_of_cores;
+
+#if defined(WEBRTC_WIN)
+  SYSTEM_INFO si;
+  GetNativeSystemInfo(&si);
+  number_of_cores = static_cast<int>(si.dwNumberOfProcessors);
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
+  number_of_cores = static_cast<int>(sysconf(_SC_NPROCESSORS_ONLN));
+  if (number_of_cores <= 0) {
+    RTC_LOG(LS_ERROR) << "Failed to get number of cores";
+    number_of_cores = 1;
+  }
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
+  int name[] = {CTL_HW, HW_AVAILCPU};
+  size_t size = sizeof(number_of_cores);
+  if (0 != sysctl(name, 2, &number_of_cores, &size, NULL, 0)) {
+    RTC_LOG(LS_ERROR) << "Failed to get number of cores";
+    number_of_cores = 1;
+  }
+#elif defined(WEBRTC_FUCHSIA)
+  number_of_cores = zx_system_get_num_cpus();
+#else
+  RTC_LOG(LS_ERROR) << "No function to get number of cores";
+  number_of_cores = 1;
+#endif
+
+  RTC_LOG(LS_INFO) << "Available number of cores: " << number_of_cores;
+
+  RTC_CHECK_GT(number_of_cores, 0);
+  return number_of_cores;
+}
+}  // namespace internal
+
+namespace webrtc {
+
+uint32_t CpuInfo::DetectNumberOfCores() {
+  // Statically cache the number of system cores available since if the process
+  // is running in a sandbox, we may only be able to read the value once (before
+  // the sandbox is initialized) and not thereafter.
+  // For more information see crbug.com/176522.
+  static const uint32_t logical_cpus =
+      static_cast<uint32_t>(internal::DetectNumberOfCores());
+  return logical_cpus;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/denormal_disabler.cc

@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 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.
+ */
+
+#include "system_wrappers/include/denormal_disabler.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+#if defined(WEBRTC_ARCH_X86_FAMILY) && defined(__clang__)
+#define WEBRTC_DENORMAL_DISABLER_X86_SUPPORTED
+#endif
+
+#if defined(WEBRTC_DENORMAL_DISABLER_X86_SUPPORTED) || \
+    defined(WEBRTC_ARCH_ARM_FAMILY)
+#define WEBRTC_DENORMAL_DISABLER_SUPPORTED
+#endif
+
+constexpr int kUnspecifiedStatusWord = -1;
+
+#if defined(WEBRTC_DENORMAL_DISABLER_SUPPORTED)
+
+// Control register bit mask to disable denormals on the hardware.
+#if defined(WEBRTC_DENORMAL_DISABLER_X86_SUPPORTED)
+// On x86 two bits are used: flush-to-zero (FTZ) and denormals-are-zero (DAZ).
+constexpr int kDenormalBitMask = 0x8040;
+#elif defined(WEBRTC_ARCH_ARM_FAMILY)
+// On ARM one bit is used: flush-to-zero (FTZ).
+constexpr int kDenormalBitMask = 1 << 24;
+#endif
+
+// Reads the relevant CPU control register and returns its value for supported
+// architectures and compilers. Otherwise returns `kUnspecifiedStatusWord`.
+int ReadStatusWord() {
+  int result = kUnspecifiedStatusWord;
+#if defined(WEBRTC_DENORMAL_DISABLER_X86_SUPPORTED)
+  asm volatile("stmxcsr %0" : "=m"(result));
+#elif defined(WEBRTC_ARCH_ARM_FAMILY) && defined(WEBRTC_ARCH_32_BITS)
+  asm volatile("vmrs %[result], FPSCR" : [result] "=r"(result));
+#elif defined(WEBRTC_ARCH_ARM_FAMILY) && defined(WEBRTC_ARCH_64_BITS)
+  asm volatile("mrs %x[result], FPCR" : [result] "=r"(result));
+#endif
+  return result;
+}
+
+// Writes `status_word` in the relevant CPU control register if the architecture
+// and the compiler are supported.
+void SetStatusWord(int status_word) {
+#if defined(WEBRTC_DENORMAL_DISABLER_X86_SUPPORTED)
+  asm volatile("ldmxcsr %0" : : "m"(status_word));
+#elif defined(WEBRTC_ARCH_ARM_FAMILY) && defined(WEBRTC_ARCH_32_BITS)
+  asm volatile("vmsr FPSCR, %[src]" : : [src] "r"(status_word));
+#elif defined(WEBRTC_ARCH_ARM_FAMILY) && defined(WEBRTC_ARCH_64_BITS)
+  asm volatile("msr FPCR, %x[src]" : : [src] "r"(status_word));
+#endif
+}
+
+// Returns true if the status word indicates that denormals are enabled.
+constexpr bool DenormalsEnabled(int status_word) {
+  return (status_word & kDenormalBitMask) != kDenormalBitMask;
+}
+
+#endif  // defined(WEBRTC_DENORMAL_DISABLER_SUPPORTED)
+
+}  // namespace
+
+#if defined(WEBRTC_DENORMAL_DISABLER_SUPPORTED)
+DenormalDisabler::DenormalDisabler() : DenormalDisabler(/*enabled=*/true) {}
+
+DenormalDisabler::DenormalDisabler(bool enabled)
+    : status_word_(enabled ? ReadStatusWord() : kUnspecifiedStatusWord),
+      disabling_activated_(enabled && DenormalsEnabled(status_word_)) {
+  if (disabling_activated_) {
+    RTC_DCHECK_NE(status_word_, kUnspecifiedStatusWord);
+    SetStatusWord(status_word_ | kDenormalBitMask);
+    RTC_DCHECK(!DenormalsEnabled(ReadStatusWord()));
+  }
+}
+
+bool DenormalDisabler::IsSupported() {
+  return true;
+}
+
+DenormalDisabler::~DenormalDisabler() {
+  if (disabling_activated_) {
+    RTC_DCHECK_NE(status_word_, kUnspecifiedStatusWord);
+    SetStatusWord(status_word_);
+  }
+}
+#else
+DenormalDisabler::DenormalDisabler() : DenormalDisabler(/*enabled=*/false) {}
+
+DenormalDisabler::DenormalDisabler(bool enabled)
+    : status_word_(kUnspecifiedStatusWord), disabling_activated_(false) {}
+
+bool DenormalDisabler::IsSupported() {
+  return false;
+}
+
+DenormalDisabler::~DenormalDisabler() = default;
+#endif
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/field_trial.cc

@@ -0,0 +1,186 @@
+// 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 "system_wrappers/include/field_trial.h"
+
+#include <stddef.h>
+
+#include <map>
+#include <string>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/string_view.h"
+#include "experiments/registered_field_trials.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_encode.h"
+
+// Simple field trial implementation, which allows client to
+// specify desired flags in InitFieldTrialsFromString.
+namespace webrtc {
+namespace field_trial {
+
+static const char* trials_init_string = NULL;
+
+namespace {
+
+constexpr char kPersistentStringSeparator = '/';
+
+flat_set<std::string>& TestKeys() {
+  static auto* test_keys = new flat_set<std::string>();
+  return *test_keys;
+}
+
+// Validates the given field trial string.
+//  E.g.:
+//    "WebRTC-experimentFoo/Enabled/WebRTC-experimentBar/Enabled100kbps/"
+//    Assigns the process to group "Enabled" on WebRTCExperimentFoo trial
+//    and to group "Enabled100kbps" on WebRTCExperimentBar.
+//
+//  E.g. invalid config:
+//    "WebRTC-experiment1/Enabled"  (note missing / separator at the end).
+bool FieldTrialsStringIsValidInternal(const absl::string_view trials) {
+  if (trials.empty())
+    return true;
+
+  size_t next_item = 0;
+  std::map<absl::string_view, absl::string_view> field_trials;
+  while (next_item < trials.length()) {
+    size_t name_end = trials.find(kPersistentStringSeparator, next_item);
+    if (name_end == trials.npos || next_item == name_end)
+      return false;
+    size_t group_name_end =
+        trials.find(kPersistentStringSeparator, name_end + 1);
+    if (group_name_end == trials.npos || name_end + 1 == group_name_end)
+      return false;
+    absl::string_view name = trials.substr(next_item, name_end - next_item);
+    absl::string_view group_name =
+        trials.substr(name_end + 1, group_name_end - name_end - 1);
+
+    next_item = group_name_end + 1;
+
+    // Fail if duplicate with different group name.
+    if (field_trials.find(name) != field_trials.end() &&
+        field_trials.find(name)->second != group_name) {
+      return false;
+    }
+
+    field_trials[name] = group_name;
+  }
+
+  return true;
+}
+
+}  // namespace
+
+bool FieldTrialsStringIsValid(absl::string_view trials_string) {
+  return FieldTrialsStringIsValidInternal(trials_string);
+}
+
+void InsertOrReplaceFieldTrialStringsInMap(
+    std::map<std::string, std::string>* fieldtrial_map,
+    const absl::string_view trials_string) {
+  if (FieldTrialsStringIsValidInternal(trials_string)) {
+    std::vector<absl::string_view> tokens = rtc::split(trials_string, '/');
+    // Skip last token which is empty due to trailing '/'.
+    for (size_t idx = 0; idx < tokens.size() - 1; idx += 2) {
+      (*fieldtrial_map)[std::string(tokens[idx])] =
+          std::string(tokens[idx + 1]);
+    }
+  } else {
+    RTC_DCHECK_NOTREACHED() << "Invalid field trials string:" << trials_string;
+  }
+}
+
+std::string MergeFieldTrialsStrings(absl::string_view first,
+                                    absl::string_view second) {
+  std::map<std::string, std::string> fieldtrial_map;
+  InsertOrReplaceFieldTrialStringsInMap(&fieldtrial_map, first);
+  InsertOrReplaceFieldTrialStringsInMap(&fieldtrial_map, second);
+
+  // Merge into fieldtrial string.
+  std::string merged = "";
+  for (auto const& fieldtrial : fieldtrial_map) {
+    merged += fieldtrial.first + '/' + fieldtrial.second + '/';
+  }
+  return merged;
+}
+
+#ifndef WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT
+std::string FindFullName(absl::string_view name) {
+#if WEBRTC_STRICT_FIELD_TRIALS == 1
+  RTC_DCHECK(absl::c_linear_search(kRegisteredFieldTrials, name) ||
+             TestKeys().contains(name))
+      << name << " is not registered, see g3doc/field-trials.md.";
+#elif WEBRTC_STRICT_FIELD_TRIALS == 2
+  RTC_LOG_IF(LS_WARNING,
+             !(absl::c_linear_search(kRegisteredFieldTrials, name) ||
+               TestKeys().contains(name)))
+      << name << " is not registered, see g3doc/field-trials.md.";
+#endif
+
+  if (trials_init_string == NULL)
+    return std::string();
+
+  absl::string_view trials_string(trials_init_string);
+  if (trials_string.empty())
+    return std::string();
+
+  size_t next_item = 0;
+  while (next_item < trials_string.length()) {
+    // Find next name/value pair in field trial configuration string.
+    size_t field_name_end =
+        trials_string.find(kPersistentStringSeparator, next_item);
+    if (field_name_end == trials_string.npos || field_name_end == next_item)
+      break;
+    size_t field_value_end =
+        trials_string.find(kPersistentStringSeparator, field_name_end + 1);
+    if (field_value_end == trials_string.npos ||
+        field_value_end == field_name_end + 1)
+      break;
+    absl::string_view field_name =
+        trials_string.substr(next_item, field_name_end - next_item);
+    absl::string_view field_value = trials_string.substr(
+        field_name_end + 1, field_value_end - field_name_end - 1);
+    next_item = field_value_end + 1;
+
+    if (name == field_name)
+      return std::string(field_value);
+  }
+  return std::string();
+}
+#endif  // WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT
+
+// Optionally initialize field trial from a string.
+void InitFieldTrialsFromString(const char* trials_string) {
+  RTC_LOG(LS_INFO) << "Setting field trial string:" << trials_string;
+  if (trials_string) {
+    RTC_DCHECK(FieldTrialsStringIsValidInternal(trials_string))
+        << "Invalid field trials string:" << trials_string;
+  };
+  trials_init_string = trials_string;
+}
+
+const char* GetFieldTrialString() {
+  return trials_init_string;
+}
+
+FieldTrialsAllowedInScopeForTesting::FieldTrialsAllowedInScopeForTesting(
+    flat_set<std::string> keys) {
+  TestKeys() = std::move(keys);
+}
+
+FieldTrialsAllowedInScopeForTesting::~FieldTrialsAllowedInScopeForTesting() {
+  TestKeys().clear();
+}
+
+}  // namespace field_trial
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/metrics.cc

@@ -0,0 +1,331 @@
+// 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 "system_wrappers/include/metrics.h"
+
+#include <algorithm>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+// Default implementation of histogram methods for WebRTC clients that do not
+// want to provide their own implementation.
+
+namespace webrtc {
+namespace metrics {
+class Histogram;
+
+namespace {
+// Limit for the maximum number of sample values that can be stored.
+// TODO(asapersson): Consider using bucket count (and set up
+// linearly/exponentially spaced buckets) if samples are logged more frequently.
+const int kMaxSampleMapSize = 300;
+
+class RtcHistogram {
+ public:
+  RtcHistogram(absl::string_view name, int min, int max, int bucket_count)
+      : min_(min), max_(max), info_(name, min, max, bucket_count) {
+    RTC_DCHECK_GT(bucket_count, 0);
+  }
+
+  RtcHistogram(const RtcHistogram&) = delete;
+  RtcHistogram& operator=(const RtcHistogram&) = delete;
+
+  void Add(int sample) {
+    sample = std::min(sample, max_);
+    sample = std::max(sample, min_ - 1);  // Underflow bucket.
+
+    MutexLock lock(&mutex_);
+    if (info_.samples.size() == kMaxSampleMapSize &&
+        info_.samples.find(sample) == info_.samples.end()) {
+      return;
+    }
+    ++info_.samples[sample];
+  }
+
+  // Returns a copy (or nullptr if there are no samples) and clears samples.
+  std::unique_ptr<SampleInfo> GetAndReset() {
+    MutexLock lock(&mutex_);
+    if (info_.samples.empty())
+      return nullptr;
+
+    SampleInfo* copy =
+        new SampleInfo(info_.name, info_.min, info_.max, info_.bucket_count);
+
+    std::swap(info_.samples, copy->samples);
+
+    return std::unique_ptr<SampleInfo>(copy);
+  }
+
+  const std::string& name() const { return info_.name; }
+
+  // Functions only for testing.
+  void Reset() {
+    MutexLock lock(&mutex_);
+    info_.samples.clear();
+  }
+
+  int NumEvents(int sample) const {
+    MutexLock lock(&mutex_);
+    const auto it = info_.samples.find(sample);
+    return (it == info_.samples.end()) ? 0 : it->second;
+  }
+
+  int NumSamples() const {
+    int num_samples = 0;
+    MutexLock lock(&mutex_);
+    for (const auto& sample : info_.samples) {
+      num_samples += sample.second;
+    }
+    return num_samples;
+  }
+
+  int MinSample() const {
+    MutexLock lock(&mutex_);
+    return (info_.samples.empty()) ? -1 : info_.samples.begin()->first;
+  }
+
+  std::map<int, int> Samples() const {
+    MutexLock lock(&mutex_);
+    return info_.samples;
+  }
+
+ private:
+  mutable Mutex mutex_;
+  const int min_;
+  const int max_;
+  SampleInfo info_ RTC_GUARDED_BY(mutex_);
+};
+
+class RtcHistogramMap {
+ public:
+  RtcHistogramMap() {}
+  ~RtcHistogramMap() {}
+
+  RtcHistogramMap(const RtcHistogramMap&) = delete;
+  RtcHistogramMap& operator=(const RtcHistogramMap&) = delete;
+
+  Histogram* GetCountsHistogram(absl::string_view name,
+                                int min,
+                                int max,
+                                int bucket_count) {
+    MutexLock lock(&mutex_);
+    const auto& it = map_.find(name);
+    if (it != map_.end())
+      return reinterpret_cast<Histogram*>(it->second.get());
+
+    RtcHistogram* hist = new RtcHistogram(name, min, max, bucket_count);
+    map_.emplace(name, hist);
+    return reinterpret_cast<Histogram*>(hist);
+  }
+
+  Histogram* GetEnumerationHistogram(absl::string_view name, int boundary) {
+    MutexLock lock(&mutex_);
+    const auto& it = map_.find(name);
+    if (it != map_.end())
+      return reinterpret_cast<Histogram*>(it->second.get());
+
+    RtcHistogram* hist = new RtcHistogram(name, 1, boundary, boundary + 1);
+    map_.emplace(name, hist);
+    return reinterpret_cast<Histogram*>(hist);
+  }
+
+  void GetAndReset(std::map<std::string,
+                            std::unique_ptr<SampleInfo>,
+                            rtc::AbslStringViewCmp>* histograms) {
+    MutexLock lock(&mutex_);
+    for (const auto& kv : map_) {
+      std::unique_ptr<SampleInfo> info = kv.second->GetAndReset();
+      if (info)
+        histograms->insert(std::make_pair(kv.first, std::move(info)));
+    }
+  }
+
+  // Functions only for testing.
+  void Reset() {
+    MutexLock lock(&mutex_);
+    for (const auto& kv : map_)
+      kv.second->Reset();
+  }
+
+  int NumEvents(absl::string_view name, int sample) const {
+    MutexLock lock(&mutex_);
+    const auto& it = map_.find(name);
+    return (it == map_.end()) ? 0 : it->second->NumEvents(sample);
+  }
+
+  int NumSamples(absl::string_view name) const {
+    MutexLock lock(&mutex_);
+    const auto& it = map_.find(name);
+    return (it == map_.end()) ? 0 : it->second->NumSamples();
+  }
+
+  int MinSample(absl::string_view name) const {
+    MutexLock lock(&mutex_);
+    const auto& it = map_.find(name);
+    return (it == map_.end()) ? -1 : it->second->MinSample();
+  }
+
+  std::map<int, int> Samples(absl::string_view name) const {
+    MutexLock lock(&mutex_);
+    const auto& it = map_.find(name);
+    return (it == map_.end()) ? std::map<int, int>() : it->second->Samples();
+  }
+
+ private:
+  mutable Mutex mutex_;
+  std::map<std::string, std::unique_ptr<RtcHistogram>, rtc::AbslStringViewCmp>
+      map_ RTC_GUARDED_BY(mutex_);
+};
+
+// RtcHistogramMap is allocated upon call to Enable().
+// The histogram getter functions, which return pointer values to the histograms
+// in the map, are cached in WebRTC. Therefore, this memory is not freed by the
+// application (the memory will be reclaimed by the OS).
+static std::atomic<RtcHistogramMap*> g_rtc_histogram_map(nullptr);
+
+void CreateMap() {
+  RtcHistogramMap* map = g_rtc_histogram_map.load(std::memory_order_acquire);
+  if (map == nullptr) {
+    RtcHistogramMap* new_map = new RtcHistogramMap();
+    if (!g_rtc_histogram_map.compare_exchange_strong(map, new_map))
+      delete new_map;
+  }
+}
+
+// Set the first time we start using histograms. Used to make sure Enable() is
+// not called thereafter.
+#if RTC_DCHECK_IS_ON
+static std::atomic<int> g_rtc_histogram_called(0);
+#endif
+
+// Gets the map (or nullptr).
+RtcHistogramMap* GetMap() {
+#if RTC_DCHECK_IS_ON
+  g_rtc_histogram_called.store(1, std::memory_order_release);
+#endif
+  return g_rtc_histogram_map.load();
+}
+}  // namespace
+
+#ifndef WEBRTC_EXCLUDE_METRICS_DEFAULT
+// Implementation of histogram methods in
+// webrtc/system_wrappers/interface/metrics.h.
+
+// Histogram with exponentially spaced buckets.
+// Creates (or finds) histogram.
+// The returned histogram pointer is cached (and used for adding samples in
+// subsequent calls).
+Histogram* HistogramFactoryGetCounts(absl::string_view name,
+                                     int min,
+                                     int max,
+                                     int bucket_count) {
+  // TODO(asapersson): Alternative implementation will be needed if this
+  // histogram type should be truly exponential.
+  return HistogramFactoryGetCountsLinear(name, min, max, bucket_count);
+}
+
+// Histogram with linearly spaced buckets.
+// Creates (or finds) histogram.
+// The returned histogram pointer is cached (and used for adding samples in
+// subsequent calls).
+Histogram* HistogramFactoryGetCountsLinear(absl::string_view name,
+                                           int min,
+                                           int max,
+                                           int bucket_count) {
+  RtcHistogramMap* map = GetMap();
+  if (!map)
+    return nullptr;
+
+  return map->GetCountsHistogram(name, min, max, bucket_count);
+}
+
+// Histogram with linearly spaced buckets.
+// Creates (or finds) histogram.
+// The returned histogram pointer is cached (and used for adding samples in
+// subsequent calls).
+Histogram* HistogramFactoryGetEnumeration(absl::string_view name,
+                                          int boundary) {
+  RtcHistogramMap* map = GetMap();
+  if (!map)
+    return nullptr;
+
+  return map->GetEnumerationHistogram(name, boundary);
+}
+
+// Our default implementation reuses the non-sparse histogram.
+Histogram* SparseHistogramFactoryGetEnumeration(absl::string_view name,
+                                                int boundary) {
+  return HistogramFactoryGetEnumeration(name, boundary);
+}
+
+// Fast path. Adds `sample` to cached `histogram_pointer`.
+void HistogramAdd(Histogram* histogram_pointer, int sample) {
+  RtcHistogram* ptr = reinterpret_cast<RtcHistogram*>(histogram_pointer);
+  ptr->Add(sample);
+}
+
+#endif  // WEBRTC_EXCLUDE_METRICS_DEFAULT
+
+SampleInfo::SampleInfo(absl::string_view name,
+                       int min,
+                       int max,
+                       size_t bucket_count)
+    : name(name), min(min), max(max), bucket_count(bucket_count) {}
+
+SampleInfo::~SampleInfo() {}
+
+// Implementation of global functions in metrics.h.
+void Enable() {
+  RTC_DCHECK(g_rtc_histogram_map.load() == nullptr);
+#if RTC_DCHECK_IS_ON
+  RTC_DCHECK_EQ(0, g_rtc_histogram_called.load(std::memory_order_acquire));
+#endif
+  CreateMap();
+}
+
+void GetAndReset(
+    std::map<std::string, std::unique_ptr<SampleInfo>, rtc::AbslStringViewCmp>*
+        histograms) {
+  histograms->clear();
+  RtcHistogramMap* map = GetMap();
+  if (map)
+    map->GetAndReset(histograms);
+}
+
+void Reset() {
+  RtcHistogramMap* map = GetMap();
+  if (map)
+    map->Reset();
+}
+
+int NumEvents(absl::string_view name, int sample) {
+  RtcHistogramMap* map = GetMap();
+  return map ? map->NumEvents(name, sample) : 0;
+}
+
+int NumSamples(absl::string_view name) {
+  RtcHistogramMap* map = GetMap();
+  return map ? map->NumSamples(name) : 0;
+}
+
+int MinSample(absl::string_view name) {
+  RtcHistogramMap* map = GetMap();
+  return map ? map->MinSample(name) : -1;
+}
+
+std::map<int, int> Samples(absl::string_view name) {
+  RtcHistogramMap* map = GetMap();
+  return map ? map->Samples(name) : std::map<int, int>();
+}
+
+}  // namespace metrics
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc

@@ -0,0 +1,153 @@
+/*
+ *  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 "system_wrappers/include/rtp_to_ntp_estimator.h"
+
+#include <stddef.h>
+
+#include <cmath>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+// Maximum number of RTCP SR reports to use to map between RTP and NTP.
+constexpr size_t kNumRtcpReportsToUse = 20;
+// Don't allow NTP timestamps to jump more than 1 hour. Chosen arbitrary as big
+// enough to not affect normal use-cases. Yet it is smaller than RTP wrap-around
+// half-period (90khz RTP clock wrap-arounds every 13.25 hours). After half of
+// wrap-around period it is impossible to unwrap RTP timestamps correctly.
+constexpr uint64_t kMaxAllowedRtcpNtpInterval = uint64_t{60 * 60} << 32;
+}  // namespace
+
+void RtpToNtpEstimator::UpdateParameters() {
+  size_t n = measurements_.size();
+  if (n < 2)
+    return;
+
+  // Run linear regression:
+  // Given x[] and y[] writes out such k and b that line y=k*x+b approximates
+  // given points in the best way (Least Squares Method).
+  auto x = [](const RtcpMeasurement& m) {
+    return static_cast<double>(m.unwrapped_rtp_timestamp);
+  };
+  auto y = [](const RtcpMeasurement& m) {
+    return static_cast<double>(static_cast<uint64_t>(m.ntp_time));
+  };
+
+  double avg_x = 0;
+  double avg_y = 0;
+  for (const RtcpMeasurement& m : measurements_) {
+    avg_x += x(m);
+    avg_y += y(m);
+  }
+  avg_x /= n;
+  avg_y /= n;
+
+  double variance_x = 0;
+  double covariance_xy = 0;
+  for (const RtcpMeasurement& m : measurements_) {
+    double normalized_x = x(m) - avg_x;
+    double normalized_y = y(m) - avg_y;
+    variance_x += normalized_x * normalized_x;
+    covariance_xy += normalized_x * normalized_y;
+  }
+
+  if (std::fabs(variance_x) < 1e-8)
+    return;
+
+  double k = covariance_xy / variance_x;
+  double b = avg_y - k * avg_x;
+  params_ = {{.slope = k, .offset = b}};
+}
+
+RtpToNtpEstimator::UpdateResult RtpToNtpEstimator::UpdateMeasurements(
+    NtpTime ntp,
+    uint32_t rtp_timestamp) {
+  int64_t unwrapped_rtp_timestamp = unwrapper_.Unwrap(rtp_timestamp);
+
+  RtcpMeasurement new_measurement = {
+      .ntp_time = ntp, .unwrapped_rtp_timestamp = unwrapped_rtp_timestamp};
+
+  for (const RtcpMeasurement& measurement : measurements_) {
+    // Use || since two equal timestamps will result in zero frequency.
+    if (measurement.ntp_time == ntp ||
+        measurement.unwrapped_rtp_timestamp == unwrapped_rtp_timestamp) {
+      return kSameMeasurement;
+    }
+  }
+
+  if (!new_measurement.ntp_time.Valid())
+    return kInvalidMeasurement;
+
+  uint64_t ntp_new = static_cast<uint64_t>(new_measurement.ntp_time);
+  bool invalid_sample = false;
+  if (!measurements_.empty()) {
+    int64_t old_rtp_timestamp = measurements_.front().unwrapped_rtp_timestamp;
+    uint64_t old_ntp = static_cast<uint64_t>(measurements_.front().ntp_time);
+    if (ntp_new <= old_ntp || ntp_new > old_ntp + kMaxAllowedRtcpNtpInterval) {
+      invalid_sample = true;
+    } else if (unwrapped_rtp_timestamp <= old_rtp_timestamp) {
+      RTC_LOG(LS_WARNING)
+          << "Newer RTCP SR report with older RTP timestamp, dropping";
+      invalid_sample = true;
+    } else if (unwrapped_rtp_timestamp - old_rtp_timestamp > (1 << 25)) {
+      // Sanity check. No jumps too far into the future in rtp.
+      invalid_sample = true;
+    }
+  }
+
+  if (invalid_sample) {
+    ++consecutive_invalid_samples_;
+    if (consecutive_invalid_samples_ < kMaxInvalidSamples) {
+      return kInvalidMeasurement;
+    }
+    RTC_LOG(LS_WARNING) << "Multiple consecutively invalid RTCP SR reports, "
+                           "clearing measurements.";
+    measurements_.clear();
+    params_ = absl::nullopt;
+  }
+  consecutive_invalid_samples_ = 0;
+
+  // Insert new RTCP SR report.
+  if (measurements_.size() == kNumRtcpReportsToUse)
+    measurements_.pop_back();
+
+  measurements_.push_front(new_measurement);
+
+  // List updated, calculate new parameters.
+  UpdateParameters();
+  return kNewMeasurement;
+}
+
+NtpTime RtpToNtpEstimator::Estimate(uint32_t rtp_timestamp) const {
+  if (!params_)
+    return NtpTime();
+
+  double estimated =
+      static_cast<double>(unwrapper_.Unwrap(rtp_timestamp)) * params_->slope +
+      params_->offset + 0.5f;
+
+  return NtpTime(rtc::saturated_cast<uint64_t>(estimated));
+}
+
+double RtpToNtpEstimator::EstimatedFrequencyKhz() const {
+  if (!params_.has_value()) {
+    return 0.0;
+  }
+  static constexpr double kNtpUnitPerMs = 4.294967296E6;  // 2^32 / 1000.
+  return kNtpUnitPerMs / params_->slope;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/system_wrappers/source/sleep.cc

@@ -0,0 +1,36 @@
+/*
+ *  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.
+ */
+// An OS-independent sleep function.
+
+#include "system_wrappers/include/sleep.h"
+
+#ifdef _WIN32
+// For Sleep()
+#include <windows.h>
+#else
+// For nanosleep()
+#include <time.h>
+#endif
+
+namespace webrtc {
+
+void SleepMs(int msecs) {
+#ifdef _WIN32
+  Sleep(msecs);
+#else
+  struct timespec short_wait;
+  struct timespec remainder;
+  short_wait.tv_sec = msecs / 1000;
+  short_wait.tv_nsec = (msecs % 1000) * 1000 * 1000;
+  nanosleep(&short_wait, &remainder);
+#endif
+}
+
+}  // namespace webrtc