Repo created

TMessagesProj/jni/voip/webrtc/audio/OWNERS

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+alessiob@webrtc.org
+gustaf@webrtc.org
+henrik.lundin@webrtc.org
+jakobi@webrtc.org
+peah@webrtc.org

TMessagesProj/jni/voip/webrtc/audio/audio_level.cc

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+/*
+ *  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 "audio/audio_level.h"
+
+#include "api/audio/audio_frame.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+namespace voe {
+
+AudioLevel::AudioLevel()
+    : abs_max_(0), count_(0), current_level_full_range_(0) {}
+
+AudioLevel::~AudioLevel() {}
+
+void AudioLevel::Reset() {
+  MutexLock lock(&mutex_);
+  abs_max_ = 0;
+  count_ = 0;
+  current_level_full_range_ = 0;
+  total_energy_ = 0.0;
+  total_duration_ = 0.0;
+}
+
+int16_t AudioLevel::LevelFullRange() const {
+  MutexLock lock(&mutex_);
+  return current_level_full_range_;
+}
+
+void AudioLevel::ResetLevelFullRange() {
+  MutexLock lock(&mutex_);
+  abs_max_ = 0;
+  count_ = 0;
+  current_level_full_range_ = 0;
+}
+
+double AudioLevel::TotalEnergy() const {
+  MutexLock lock(&mutex_);
+  return total_energy_;
+}
+
+double AudioLevel::TotalDuration() const {
+  MutexLock lock(&mutex_);
+  return total_duration_;
+}
+
+void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
+  // Check speech level (works for 2 channels as well)
+  int16_t abs_value =
+      audioFrame.muted()
+          ? 0
+          : WebRtcSpl_MaxAbsValueW16(
+                audioFrame.data(),
+                audioFrame.samples_per_channel_ * audioFrame.num_channels_);
+
+  // Protect member access using a lock since this method is called on a
+  // dedicated audio thread in the RecordedDataIsAvailable() callback.
+  MutexLock lock(&mutex_);
+
+  if (abs_value > abs_max_)
+    abs_max_ = abs_value;
+
+  // Update level approximately 9 times per second, assuming audio frame
+  // duration is approximately 10 ms. (The update frequency is every
+  // 11th (= |kUpdateFrequency+1|) call: 1000/(11*10)=9.09..., we should
+  // probably change this behavior, see https://crbug.com/webrtc/10784).
+  if (count_++ == kUpdateFrequency) {
+    current_level_full_range_ = abs_max_;
+
+    count_ = 0;
+
+    // Decay the absolute maximum (divide by 4)
+    abs_max_ >>= 2;
+  }
+
+  // See the description for "totalAudioEnergy" in the WebRTC stats spec
+  // (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
+  // for an explanation of these formulas. In short, we need a value that can
+  // be used to compute RMS audio levels over different time intervals, by
+  // taking the difference between the results from two getStats calls. To do
+  // this, the value needs to be of units "squared sample value * time".
+  double additional_energy =
+      static_cast<double>(current_level_full_range_) / INT16_MAX;
+  additional_energy *= additional_energy;
+  total_energy_ += additional_energy * duration;
+  total_duration_ += duration;
+}
+
+}  // namespace voe
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/audio_level.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 AUDIO_AUDIO_LEVEL_H_
+#define AUDIO_AUDIO_LEVEL_H_
+
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioFrame;
+namespace voe {
+
+// This class is thread-safe. However, TotalEnergy() and TotalDuration() are
+// related, so if you call ComputeLevel() on a different thread than you read
+// these values, you still need to use lock to read them as a pair.
+class AudioLevel {
+ public:
+  AudioLevel();
+  ~AudioLevel();
+  void Reset();
+
+  // Returns the current audio level linearly [0,32767], which gets updated
+  // every "kUpdateFrequency+1" call to ComputeLevel() based on the maximum
+  // audio level of any audio frame, decaying by a factor of 1/4 each time
+  // LevelFullRange() gets updated.
+  // Called on "API thread(s)" from APIs like VoEBase::CreateChannel(),
+  // VoEBase::StopSend().
+  int16_t LevelFullRange() const;
+  void ResetLevelFullRange();
+  // See the description for "totalAudioEnergy" in the WebRTC stats spec
+  // (https://w3c.github.io/webrtc-stats/#dom-rtcaudiohandlerstats-totalaudioenergy)
+  // In our implementation, the total audio energy increases by the
+  // energy-equivalent of LevelFullRange() at the time of ComputeLevel(), rather
+  // than the energy of the samples in that specific audio frame. As a result,
+  // we may report a higher audio energy and audio level than the spec mandates.
+  // TODO(https://crbug.com/webrtc/10784): We should either do what the spec
+  // says or update the spec to match our implementation. If we want to have a
+  // decaying audio level we should probably update both the spec and the
+  // implementation to reduce the complexity of the definition. If we want to
+  // continue to have decaying audio we should have unittests covering the
+  // behavior of the decay.
+  double TotalEnergy() const;
+  double TotalDuration() const;
+
+  // Called on a native capture audio thread (platform dependent) from the
+  // AudioTransport::RecordedDataIsAvailable() callback.
+  // In Chrome, this method is called on the AudioInputDevice thread.
+  void ComputeLevel(const AudioFrame& audioFrame, double duration);
+
+ private:
+  enum { kUpdateFrequency = 10 };
+
+  mutable Mutex mutex_;
+
+  int16_t abs_max_ RTC_GUARDED_BY(mutex_);
+  int16_t count_ RTC_GUARDED_BY(mutex_);
+  int16_t current_level_full_range_ RTC_GUARDED_BY(mutex_);
+
+  double total_energy_ RTC_GUARDED_BY(mutex_) = 0.0;
+  double total_duration_ RTC_GUARDED_BY(mutex_) = 0.0;
+};
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_LEVEL_H_

TMessagesProj/jni/voip/webrtc/audio/audio_receive_stream.cc

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+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_receive_stream.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/audio_sink.h"
+#include "api/rtp_parameters.h"
+#include "api/sequence_checker.h"
+#include "audio/audio_send_stream.h"
+#include "audio/audio_state.h"
+#include "audio/channel_receive.h"
+#include "audio/conversion.h"
+#include "call/rtp_config.h"
+#include "call/rtp_stream_receiver_controller_interface.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "{remote_ssrc: " << remote_ssrc;
+  ss << ", local_ssrc: " << local_ssrc;
+  ss << ", nack: " << nack.ToString();
+  ss << '}';
+  return ss.str();
+}
+
+std::string AudioReceiveStreamInterface::Config::ToString() const {
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "{rtp: " << rtp.ToString();
+  ss << ", rtcp_send_transport: "
+     << (rtcp_send_transport ? "(Transport)" : "null");
+  if (!sync_group.empty()) {
+    ss << ", sync_group: " << sync_group;
+  }
+  ss << '}';
+  return ss.str();
+}
+
+namespace {
+std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
+    Clock* clock,
+    webrtc::AudioState* audio_state,
+    NetEqFactory* neteq_factory,
+    const webrtc::AudioReceiveStreamInterface::Config& config,
+    RtcEventLog* event_log) {
+  RTC_DCHECK(audio_state);
+  internal::AudioState* internal_audio_state =
+      static_cast<internal::AudioState*>(audio_state);
+  return voe::CreateChannelReceive(
+      clock, neteq_factory, internal_audio_state->audio_device_module(),
+      config.rtcp_send_transport, event_log, config.rtp.local_ssrc,
+      config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
+      config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
+      config.enable_non_sender_rtt, config.decoder_factory,
+      config.codec_pair_id, std::move(config.frame_decryptor),
+      config.crypto_options, std::move(config.frame_transformer));
+}
+}  // namespace
+
+AudioReceiveStreamImpl::AudioReceiveStreamImpl(
+    Clock* clock,
+    PacketRouter* packet_router,
+    NetEqFactory* neteq_factory,
+    const webrtc::AudioReceiveStreamInterface::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    webrtc::RtcEventLog* event_log)
+    : AudioReceiveStreamImpl(clock,
+                             packet_router,
+                             config,
+                             audio_state,
+                             event_log,
+                             CreateChannelReceive(clock,
+                                                  audio_state.get(),
+                                                  neteq_factory,
+                                                  config,
+                                                  event_log)) {}
+
+AudioReceiveStreamImpl::AudioReceiveStreamImpl(
+    Clock* clock,
+    PacketRouter* packet_router,
+    const webrtc::AudioReceiveStreamInterface::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    webrtc::RtcEventLog* event_log,
+    std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
+    : config_(config),
+      audio_state_(audio_state),
+      source_tracker_(clock),
+      channel_receive_(std::move(channel_receive)) {
+  RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl: " << config.rtp.remote_ssrc;
+  RTC_DCHECK(config.decoder_factory);
+  RTC_DCHECK(config.rtcp_send_transport);
+  RTC_DCHECK(audio_state_);
+  RTC_DCHECK(channel_receive_);
+
+  RTC_DCHECK(packet_router);
+  // Configure bandwidth estimation.
+  channel_receive_->RegisterReceiverCongestionControlObjects(packet_router);
+
+  // When output is muted, ChannelReceive will directly notify the source
+  // tracker of "delivered" frames, so RtpReceiver information will continue to
+  // be updated.
+  channel_receive_->SetSourceTracker(&source_tracker_);
+
+  // Complete configuration.
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  channel_receive_->SetNACKStatus(config.rtp.nack.rtp_history_ms != 0,
+                                  config.rtp.nack.rtp_history_ms / 20);
+  channel_receive_->SetReceiveCodecs(config.decoder_map);
+  // `frame_transformer` and `frame_decryptor` have been given to
+  // `channel_receive_` already.
+}
+
+AudioReceiveStreamImpl::~AudioReceiveStreamImpl() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_LOG(LS_INFO) << "~AudioReceiveStreamImpl: " << remote_ssrc();
+  Stop();
+  channel_receive_->SetAssociatedSendChannel(nullptr);
+  channel_receive_->ResetReceiverCongestionControlObjects();
+}
+
+void AudioReceiveStreamImpl::RegisterWithTransport(
+    RtpStreamReceiverControllerInterface* receiver_controller) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  RTC_DCHECK(!rtp_stream_receiver_);
+  rtp_stream_receiver_ = receiver_controller->CreateReceiver(
+      remote_ssrc(), channel_receive_.get());
+}
+
+void AudioReceiveStreamImpl::UnregisterFromTransport() {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  rtp_stream_receiver_.reset();
+}
+
+void AudioReceiveStreamImpl::ReconfigureForTesting(
+    const webrtc::AudioReceiveStreamInterface::Config& config) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+
+  // SSRC can't be changed mid-stream.
+  RTC_DCHECK_EQ(remote_ssrc(), config.rtp.remote_ssrc);
+  RTC_DCHECK_EQ(local_ssrc(), config.rtp.local_ssrc);
+
+  // Configuration parameters which cannot be changed.
+  RTC_DCHECK_EQ(config_.rtcp_send_transport, config.rtcp_send_transport);
+  // Decoder factory cannot be changed because it is configured at
+  // voe::Channel construction time.
+  RTC_DCHECK_EQ(config_.decoder_factory, config.decoder_factory);
+
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  RTC_DCHECK_EQ(config_.rtp.nack.rtp_history_ms, config.rtp.nack.rtp_history_ms)
+      << "Use SetUseTransportCcAndNackHistory";
+
+  RTC_DCHECK(config_.decoder_map == config.decoder_map) << "Use SetDecoderMap";
+  RTC_DCHECK_EQ(config_.frame_transformer, config.frame_transformer)
+      << "Use SetDepacketizerToDecoderFrameTransformer";
+
+  config_ = config;
+}
+
+void AudioReceiveStreamImpl::Start() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (playing_) {
+    return;
+  }
+  RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Start: " << remote_ssrc();
+  channel_receive_->StartPlayout();
+  playing_ = true;
+  audio_state()->AddReceivingStream(this);
+}
+
+void AudioReceiveStreamImpl::Stop() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!playing_) {
+    return;
+  }
+  RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl::Stop: " << remote_ssrc();
+  channel_receive_->StopPlayout();
+  playing_ = false;
+  audio_state()->RemoveReceivingStream(this);
+}
+
+bool AudioReceiveStreamImpl::IsRunning() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return playing_;
+}
+
+void AudioReceiveStreamImpl::SetDepacketizerToDecoderFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetDepacketizerToDecoderFrameTransformer(
+      std::move(frame_transformer));
+}
+
+void AudioReceiveStreamImpl::SetDecoderMap(
+    std::map<int, SdpAudioFormat> decoder_map) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  config_.decoder_map = std::move(decoder_map);
+  channel_receive_->SetReceiveCodecs(config_.decoder_map);
+}
+
+void AudioReceiveStreamImpl::SetNackHistory(int history_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_GE(history_ms, 0);
+
+  if (config_.rtp.nack.rtp_history_ms == history_ms)
+    return;
+
+  config_.rtp.nack.rtp_history_ms = history_ms;
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  channel_receive_->SetNACKStatus(history_ms != 0, history_ms / 20);
+}
+
+void AudioReceiveStreamImpl::SetNonSenderRttMeasurement(bool enabled) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  config_.enable_non_sender_rtt = enabled;
+  channel_receive_->SetNonSenderRttMeasurement(enabled);
+}
+
+void AudioReceiveStreamImpl::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  // TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream,
+  // expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetFrameDecryptor(std::move(frame_decryptor));
+}
+
+webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
+    bool get_and_clear_legacy_stats) const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  webrtc::AudioReceiveStreamInterface::Stats stats;
+  stats.remote_ssrc = remote_ssrc();
+
+  webrtc::CallReceiveStatistics call_stats =
+      channel_receive_->GetRTCPStatistics();
+  // TODO(solenberg): Don't return here if we can't get the codec - return the
+  //                  stats we *can* get.
+  auto receive_codec = channel_receive_->GetReceiveCodec();
+  if (!receive_codec) {
+    return stats;
+  }
+
+  stats.payload_bytes_received = call_stats.payload_bytes_received;
+  stats.header_and_padding_bytes_received =
+      call_stats.header_and_padding_bytes_received;
+  stats.packets_received = call_stats.packetsReceived;
+  stats.packets_lost = call_stats.cumulativeLost;
+  stats.nacks_sent = call_stats.nacks_sent;
+  stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
+  stats.last_packet_received = call_stats.last_packet_received;
+  stats.codec_name = receive_codec->second.name;
+  stats.codec_payload_type = receive_codec->first;
+  int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
+  if (clockrate_khz > 0) {
+    stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;
+  }
+  stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
+  stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
+  stats.total_output_energy = channel_receive_->GetTotalOutputEnergy();
+  stats.total_output_duration = channel_receive_->GetTotalOutputDuration();
+  stats.estimated_playout_ntp_timestamp_ms =
+      channel_receive_->GetCurrentEstimatedPlayoutNtpTimestampMs(
+          rtc::TimeMillis());
+
+  // Get jitter buffer and total delay (alg + jitter + playout) stats.
+  auto ns = channel_receive_->GetNetworkStatistics(get_and_clear_legacy_stats);
+  stats.packets_discarded = ns.packetsDiscarded;
+  stats.fec_packets_received = ns.fecPacketsReceived;
+  stats.fec_packets_discarded = ns.fecPacketsDiscarded;
+  stats.jitter_buffer_ms = ns.currentBufferSize;
+  stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
+  stats.total_samples_received = ns.totalSamplesReceived;
+  stats.concealed_samples = ns.concealedSamples;
+  stats.silent_concealed_samples = ns.silentConcealedSamples;
+  stats.concealment_events = ns.concealmentEvents;
+  stats.jitter_buffer_delay_seconds =
+      static_cast<double>(ns.jitterBufferDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.jitter_buffer_emitted_count = ns.jitterBufferEmittedCount;
+  stats.jitter_buffer_target_delay_seconds =
+      static_cast<double>(ns.jitterBufferTargetDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.jitter_buffer_minimum_delay_seconds =
+      static_cast<double>(ns.jitterBufferMinimumDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.inserted_samples_for_deceleration = ns.insertedSamplesForDeceleration;
+  stats.removed_samples_for_acceleration = ns.removedSamplesForAcceleration;
+  stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
+  stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
+  stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
+  stats.secondary_discarded_rate = Q14ToFloat(ns.currentSecondaryDiscardedRate);
+  stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
+  stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
+  stats.jitter_buffer_flushes = ns.packetBufferFlushes;
+  stats.delayed_packet_outage_samples = ns.delayedPacketOutageSamples;
+  stats.relative_packet_arrival_delay_seconds =
+      static_cast<double>(ns.relativePacketArrivalDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.interruption_count = ns.interruptionCount;
+  stats.total_interruption_duration_ms = ns.totalInterruptionDurationMs;
+
+  auto ds = channel_receive_->GetDecodingCallStatistics();
+  stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
+  stats.decoding_calls_to_neteq = ds.calls_to_neteq;
+  stats.decoding_normal = ds.decoded_normal;
+  stats.decoding_plc = ds.decoded_neteq_plc;
+  stats.decoding_codec_plc = ds.decoded_codec_plc;
+  stats.decoding_cng = ds.decoded_cng;
+  stats.decoding_plc_cng = ds.decoded_plc_cng;
+  stats.decoding_muted_output = ds.decoded_muted_output;
+
+  stats.last_sender_report_timestamp_ms =
+      call_stats.last_sender_report_timestamp_ms;
+  stats.last_sender_report_remote_timestamp_ms =
+      call_stats.last_sender_report_remote_timestamp_ms;
+  stats.sender_reports_packets_sent = call_stats.sender_reports_packets_sent;
+  stats.sender_reports_bytes_sent = call_stats.sender_reports_bytes_sent;
+  stats.sender_reports_reports_count = call_stats.sender_reports_reports_count;
+  stats.round_trip_time = call_stats.round_trip_time;
+  stats.round_trip_time_measurements = call_stats.round_trip_time_measurements;
+  stats.total_round_trip_time = call_stats.total_round_trip_time;
+
+  return stats;
+}
+
+void AudioReceiveStreamImpl::SetSink(AudioSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetSink(sink);
+}
+
+void AudioReceiveStreamImpl::SetGain(float gain) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetChannelOutputVolumeScaling(gain);
+}
+
+bool AudioReceiveStreamImpl::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->SetBaseMinimumPlayoutDelayMs(delay_ms);
+}
+
+int AudioReceiveStreamImpl::GetBaseMinimumPlayoutDelayMs() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->GetBaseMinimumPlayoutDelayMs();
+}
+
+std::vector<RtpSource> AudioReceiveStreamImpl::GetSources() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return source_tracker_.GetSources();
+}
+
+AudioMixer::Source::AudioFrameInfo
+AudioReceiveStreamImpl::GetAudioFrameWithInfo(int sample_rate_hz,
+                                              AudioFrame* audio_frame) {
+  AudioMixer::Source::AudioFrameInfo audio_frame_info =
+      channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+  if (audio_frame_info != AudioMixer::Source::AudioFrameInfo::kError &&
+      !audio_frame->packet_infos_.empty()) {
+    source_tracker_.OnFrameDelivered(audio_frame->packet_infos_);
+  }
+  return audio_frame_info;
+}
+
+int AudioReceiveStreamImpl::Ssrc() const {
+  return remote_ssrc();
+}
+
+int AudioReceiveStreamImpl::PreferredSampleRate() const {
+  return channel_receive_->PreferredSampleRate();
+}
+
+uint32_t AudioReceiveStreamImpl::id() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return remote_ssrc();
+}
+
+absl::optional<Syncable::Info> AudioReceiveStreamImpl::GetInfo() const {
+  // TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer,
+  // expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->GetSyncInfo();
+}
+
+bool AudioReceiveStreamImpl::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                                                    int64_t* time_ms) const {
+  // Called on video capture thread.
+  return channel_receive_->GetPlayoutRtpTimestamp(rtp_timestamp, time_ms);
+}
+
+void AudioReceiveStreamImpl::SetEstimatedPlayoutNtpTimestampMs(
+    int64_t ntp_timestamp_ms,
+    int64_t time_ms) {
+  // Called on video capture thread.
+  channel_receive_->SetEstimatedPlayoutNtpTimestampMs(ntp_timestamp_ms,
+                                                      time_ms);
+}
+
+bool AudioReceiveStreamImpl::SetMinimumPlayoutDelay(int delay_ms) {
+  // TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer,
+  // expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->SetMinimumPlayoutDelay(delay_ms);
+}
+
+void AudioReceiveStreamImpl::AssociateSendStream(
+    internal::AudioSendStream* send_stream) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  channel_receive_->SetAssociatedSendChannel(
+      send_stream ? send_stream->GetChannel() : nullptr);
+  associated_send_stream_ = send_stream;
+}
+
+void AudioReceiveStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.IsCurrent());
+  channel_receive_->ReceivedRTCPPacket(packet, length);
+}
+
+void AudioReceiveStreamImpl::SetSyncGroup(absl::string_view sync_group) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  config_.sync_group = std::string(sync_group);
+}
+
+void AudioReceiveStreamImpl::SetLocalSsrc(uint32_t local_ssrc) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  // TODO(tommi): Consider storing local_ssrc in one place.
+  config_.rtp.local_ssrc = local_ssrc;
+  channel_receive_->OnLocalSsrcChange(local_ssrc);
+}
+
+uint32_t AudioReceiveStreamImpl::local_ssrc() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  RTC_DCHECK_EQ(config_.rtp.local_ssrc, channel_receive_->GetLocalSsrc());
+  return config_.rtp.local_ssrc;
+}
+
+const std::string& AudioReceiveStreamImpl::sync_group() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return config_.sync_group;
+}
+
+const AudioSendStream*
+AudioReceiveStreamImpl::GetAssociatedSendStreamForTesting() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return associated_send_stream_;
+}
+
+internal::AudioState* AudioReceiveStreamImpl::audio_state() const {
+  auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/audio_receive_stream.h

@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_RECEIVE_STREAM_H_
+#define AUDIO_AUDIO_RECEIVE_STREAM_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_mixer.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/rtp_headers.h"
+#include "api/sequence_checker.h"
+#include "audio/audio_state.h"
+#include "call/audio_receive_stream.h"
+#include "call/syncable.h"
+#include "modules/rtp_rtcp/source/source_tracker.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class PacketRouter;
+class RtcEventLog;
+class RtpStreamReceiverControllerInterface;
+class RtpStreamReceiverInterface;
+
+namespace voe {
+class ChannelReceiveInterface;
+}  // namespace voe
+
+namespace internal {
+class AudioSendStream;
+}  // namespace internal
+
+class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface,
+                                     public AudioMixer::Source,
+                                     public Syncable {
+ public:
+  AudioReceiveStreamImpl(
+      Clock* clock,
+      PacketRouter* packet_router,
+      NetEqFactory* neteq_factory,
+      const webrtc::AudioReceiveStreamInterface::Config& config,
+      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+      webrtc::RtcEventLog* event_log);
+  // For unit tests, which need to supply a mock channel receive.
+  AudioReceiveStreamImpl(
+      Clock* clock,
+      PacketRouter* packet_router,
+      const webrtc::AudioReceiveStreamInterface::Config& config,
+      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+      webrtc::RtcEventLog* event_log,
+      std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
+
+  AudioReceiveStreamImpl() = delete;
+  AudioReceiveStreamImpl(const AudioReceiveStreamImpl&) = delete;
+  AudioReceiveStreamImpl& operator=(const AudioReceiveStreamImpl&) = delete;
+
+  // Destruction happens on the worker thread. Prior to destruction the caller
+  // must ensure that a registration with the transport has been cleared. See
+  // `RegisterWithTransport` for details.
+  // TODO(tommi): As a further improvement to this, performing the full
+  // destruction on the network thread could be made the default.
+  ~AudioReceiveStreamImpl() override;
+
+  // Called on the network thread to register/unregister with the network
+  // transport.
+  void RegisterWithTransport(
+      RtpStreamReceiverControllerInterface* receiver_controller);
+  // If registration has previously been done (via `RegisterWithTransport`) then
+  // `UnregisterFromTransport` must be called prior to destruction, on the
+  // network thread.
+  void UnregisterFromTransport();
+
+  // webrtc::AudioReceiveStreamInterface implementation.
+  void Start() override;
+  void Stop() override;
+  bool IsRunning() const override;
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+  void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
+  void SetNackHistory(int history_ms) override;
+  void SetNonSenderRttMeasurement(bool enabled) override;
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+
+  webrtc::AudioReceiveStreamInterface::Stats GetStats(
+      bool get_and_clear_legacy_stats) const override;
+  void SetSink(AudioSinkInterface* sink) override;
+  void SetGain(float gain) override;
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
+  int GetBaseMinimumPlayoutDelayMs() const override;
+  std::vector<webrtc::RtpSource> GetSources() const override;
+
+  // AudioMixer::Source
+  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                       AudioFrame* audio_frame) override;
+  int Ssrc() const override;
+  int PreferredSampleRate() const override;
+
+  // Syncable
+  uint32_t id() const override;
+  absl::optional<Syncable::Info> GetInfo() const override;
+  bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                              int64_t* time_ms) const override;
+  void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                         int64_t time_ms) override;
+  bool SetMinimumPlayoutDelay(int delay_ms) override;
+
+  void AssociateSendStream(internal::AudioSendStream* send_stream);
+  void DeliverRtcp(const uint8_t* packet, size_t length);
+
+  void SetSyncGroup(absl::string_view sync_group);
+
+  void SetLocalSsrc(uint32_t local_ssrc);
+
+  uint32_t local_ssrc() const;
+
+  uint32_t remote_ssrc() const override {
+    // The remote_ssrc member variable of config_ will never change and can be
+    // considered const.
+    return config_.rtp.remote_ssrc;
+  }
+
+  // Returns a reference to the currently set sync group of the stream.
+  // Must be called on the packet delivery thread.
+  const std::string& sync_group() const;
+
+  const AudioSendStream* GetAssociatedSendStreamForTesting() const;
+
+  // TODO(tommi): Remove this method.
+  void ReconfigureForTesting(
+      const webrtc::AudioReceiveStreamInterface::Config& config);
+
+ private:
+  internal::AudioState* audio_state() const;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
+  // TODO(bugs.webrtc.org/11993): This checker conceptually represents
+  // operations that belong to the network thread. The Call class is currently
+  // moving towards handling network packets on the network thread and while
+  // that work is ongoing, this checker may in practice represent the worker
+  // thread, but still serves as a mechanism of grouping together concepts
+  // that belong to the network thread. Once the packets are fully delivered
+  // on the network thread, this comment will be deleted.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_{
+      SequenceChecker::kDetached};
+  webrtc::AudioReceiveStreamInterface::Config config_;
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  SourceTracker source_tracker_;
+  const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
+  AudioSendStream* associated_send_stream_
+      RTC_GUARDED_BY(packet_sequence_checker_) = nullptr;
+
+  bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
+
+  std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_
+      RTC_GUARDED_BY(packet_sequence_checker_);
+};
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_RECEIVE_STREAM_H_

TMessagesProj/jni/voip/webrtc/audio/audio_send_stream.cc

@@ -0,0 +1,891 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_send_stream.h"
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/transport.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/function_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/task_queue/task_queue_base.h"
+#include "audio/audio_state.h"
+#include "audio/channel_send.h"
+#include "audio/conversion.h"
+#include "call/rtp_config.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "common_audio/vad/include/vad.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
+#include "logging/rtc_event_log/rtc_stream_config.h"
+#include "media/base/media_channel.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+namespace {
+
+void UpdateEventLogStreamConfig(RtcEventLog* event_log,
+                                const AudioSendStream::Config& config,
+                                const AudioSendStream::Config* old_config) {
+  using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
+  // Only update if any of the things we log have changed.
+  auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
+                                const absl::optional<SendCodecSpec>& b) {
+    if (a.has_value() && b.has_value()) {
+      return a->format.name == b->format.name &&
+             a->payload_type == b->payload_type;
+    }
+    return !a.has_value() && !b.has_value();
+  };
+
+  if (old_config && config.rtp.ssrc == old_config->rtp.ssrc &&
+      config.rtp.extensions == old_config->rtp.extensions &&
+      payload_types_equal(config.send_codec_spec,
+                          old_config->send_codec_spec)) {
+    return;
+  }
+
+  auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
+  rtclog_config->local_ssrc = config.rtp.ssrc;
+  rtclog_config->rtp_extensions = config.rtp.extensions;
+  if (config.send_codec_spec) {
+    rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
+                                       config.send_codec_spec->payload_type, 0);
+  }
+  event_log->Log(std::make_unique<RtcEventAudioSendStreamConfig>(
+      std::move(rtclog_config)));
+}
+
+}  // namespace
+
+constexpr char AudioAllocationConfig::kKey[];
+
+std::unique_ptr<StructParametersParser> AudioAllocationConfig::Parser() {
+  return StructParametersParser::Create(       //
+      "min", &min_bitrate,                     //
+      "max", &max_bitrate,                     //
+      "prio_rate", &priority_bitrate,          //
+      "prio_rate_raw", &priority_bitrate_raw,  //
+      "rate_prio", &bitrate_priority);
+}
+
+AudioAllocationConfig::AudioAllocationConfig(
+    const FieldTrialsView& field_trials) {
+  Parser()->Parse(field_trials.Lookup(kKey));
+  if (priority_bitrate_raw && !priority_bitrate.IsZero()) {
+    RTC_LOG(LS_WARNING) << "'priority_bitrate' and '_raw' are mutually "
+                           "exclusive but both were configured.";
+  }
+}
+
+namespace internal {
+AudioSendStream::AudioSendStream(
+    Clock* clock,
+    const webrtc::AudioSendStream::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    TaskQueueFactory* task_queue_factory,
+    RtpTransportControllerSendInterface* rtp_transport,
+    BitrateAllocatorInterface* bitrate_allocator,
+    RtcEventLog* event_log,
+    RtcpRttStats* rtcp_rtt_stats,
+    const absl::optional<RtpState>& suspended_rtp_state,
+    const FieldTrialsView& field_trials)
+    : AudioSendStream(clock,
+                      config,
+                      audio_state,
+                      task_queue_factory,
+                      rtp_transport,
+                      bitrate_allocator,
+                      event_log,
+                      suspended_rtp_state,
+                      voe::CreateChannelSend(clock,
+                                             task_queue_factory,
+                                             config.send_transport,
+                                             rtcp_rtt_stats,
+                                             event_log,
+                                             config.frame_encryptor.get(),
+                                             config.crypto_options,
+                                             config.rtp.extmap_allow_mixed,
+                                             config.rtcp_report_interval_ms,
+                                             config.rtp.ssrc,
+                                             config.frame_transformer,
+                                             rtp_transport,
+                                             field_trials),
+                      field_trials) {}
+
+AudioSendStream::AudioSendStream(
+    Clock* clock,
+    const webrtc::AudioSendStream::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    TaskQueueFactory* task_queue_factory,
+    RtpTransportControllerSendInterface* rtp_transport,
+    BitrateAllocatorInterface* bitrate_allocator,
+    RtcEventLog* event_log,
+    const absl::optional<RtpState>& suspended_rtp_state,
+    std::unique_ptr<voe::ChannelSendInterface> channel_send,
+    const FieldTrialsView& field_trials)
+    : clock_(clock),
+      field_trials_(field_trials),
+      allocate_audio_without_feedback_(
+          field_trials_.IsEnabled("WebRTC-Audio-ABWENoTWCC")),
+      enable_audio_alr_probing_(
+          !field_trials_.IsDisabled("WebRTC-Audio-AlrProbing")),
+      allocation_settings_(field_trials_),
+      config_(Config(/*send_transport=*/nullptr)),
+      audio_state_(audio_state),
+      channel_send_(std::move(channel_send)),
+      event_log_(event_log),
+      use_legacy_overhead_calculation_(
+          field_trials_.IsEnabled("WebRTC-Audio-LegacyOverhead")),
+      enable_priority_bitrate_(
+          !field_trials_.IsDisabled("WebRTC-Audio-PriorityBitrate")),
+      bitrate_allocator_(bitrate_allocator),
+      rtp_transport_(rtp_transport),
+      rtp_rtcp_module_(channel_send_->GetRtpRtcp()),
+      suspended_rtp_state_(suspended_rtp_state) {
+  RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc;
+  RTC_DCHECK(audio_state_);
+  RTC_DCHECK(channel_send_);
+  RTC_DCHECK(bitrate_allocator_);
+  RTC_DCHECK(rtp_transport);
+
+  RTC_DCHECK(rtp_rtcp_module_);
+
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  ConfigureStream(config, true, nullptr);
+}
+
+AudioSendStream::~AudioSendStream() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc;
+  RTC_DCHECK(!sending_);
+  channel_send_->ResetSenderCongestionControlObjects();
+}
+
+const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_;
+}
+
+void AudioSendStream::Reconfigure(
+    const webrtc::AudioSendStream::Config& new_config,
+    SetParametersCallback callback) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  ConfigureStream(new_config, false, std::move(callback));
+}
+
+AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
+    const std::vector<RtpExtension>& extensions) {
+  ExtensionIds ids;
+  for (const auto& extension : extensions) {
+    if (extension.uri == RtpExtension::kAudioLevelUri) {
+      ids.audio_level = extension.id;
+    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
+      ids.abs_send_time = extension.id;
+    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
+      ids.transport_sequence_number = extension.id;
+    } else if (extension.uri == RtpExtension::kMidUri) {
+      ids.mid = extension.id;
+    } else if (extension.uri == RtpExtension::kRidUri) {
+      ids.rid = extension.id;
+    } else if (extension.uri == RtpExtension::kRepairedRidUri) {
+      ids.repaired_rid = extension.id;
+    } else if (extension.uri == RtpExtension::kAbsoluteCaptureTimeUri) {
+      ids.abs_capture_time = extension.id;
+    }
+  }
+  return ids;
+}
+
+int AudioSendStream::TransportSeqNumId(const AudioSendStream::Config& config) {
+  return FindExtensionIds(config.rtp.extensions).transport_sequence_number;
+}
+
+void AudioSendStream::ConfigureStream(
+    const webrtc::AudioSendStream::Config& new_config,
+    bool first_time,
+    SetParametersCallback callback) {
+  RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
+                   << new_config.ToString();
+  UpdateEventLogStreamConfig(event_log_, new_config,
+                             first_time ? nullptr : &config_);
+
+  const auto& old_config = config_;
+
+  // Configuration parameters which cannot be changed.
+  RTC_DCHECK(first_time ||
+             old_config.send_transport == new_config.send_transport);
+  RTC_DCHECK(first_time || old_config.rtp.ssrc == new_config.rtp.ssrc);
+  if (suspended_rtp_state_ && first_time) {
+    rtp_rtcp_module_->SetRtpState(*suspended_rtp_state_);
+  }
+  if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
+    channel_send_->SetRTCP_CNAME(new_config.rtp.c_name);
+  }
+
+  // Enable the frame encryptor if a new frame encryptor has been provided.
+  if (first_time || new_config.frame_encryptor != old_config.frame_encryptor) {
+    channel_send_->SetFrameEncryptor(new_config.frame_encryptor);
+  }
+
+  if (first_time ||
+      new_config.frame_transformer != old_config.frame_transformer) {
+    channel_send_->SetEncoderToPacketizerFrameTransformer(
+        new_config.frame_transformer);
+  }
+
+  if (first_time ||
+      new_config.rtp.extmap_allow_mixed != old_config.rtp.extmap_allow_mixed) {
+    rtp_rtcp_module_->SetExtmapAllowMixed(new_config.rtp.extmap_allow_mixed);
+  }
+
+  const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions);
+  const ExtensionIds new_ids = FindExtensionIds(new_config.rtp.extensions);
+
+  // Audio level indication
+  if (first_time || new_ids.audio_level != old_ids.audio_level) {
+    channel_send_->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
+                                                     new_ids.audio_level);
+  }
+
+  if (first_time || new_ids.abs_send_time != old_ids.abs_send_time) {
+    absl::string_view uri = AbsoluteSendTime::Uri();
+    rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(uri);
+    if (new_ids.abs_send_time) {
+      rtp_rtcp_module_->RegisterRtpHeaderExtension(uri, new_ids.abs_send_time);
+    }
+  }
+
+  bool transport_seq_num_id_changed =
+      new_ids.transport_sequence_number != old_ids.transport_sequence_number;
+  if (first_time ||
+      (transport_seq_num_id_changed && !allocate_audio_without_feedback_)) {
+    if (!first_time) {
+      channel_send_->ResetSenderCongestionControlObjects();
+    }
+
+    if (!allocate_audio_without_feedback_ &&
+        new_ids.transport_sequence_number != 0) {
+      rtp_rtcp_module_->RegisterRtpHeaderExtension(
+          TransportSequenceNumber::Uri(), new_ids.transport_sequence_number);
+      // Probing in application limited region is only used in combination with
+      // send side congestion control, wich depends on feedback packets which
+      // requires transport sequence numbers to be enabled.
+      // Optionally request ALR probing but do not override any existing
+      // request from other streams.
+      if (enable_audio_alr_probing_) {
+        rtp_transport_->EnablePeriodicAlrProbing(true);
+      }
+    }
+    channel_send_->RegisterSenderCongestionControlObjects(rtp_transport_);
+  }
+  // MID RTP header extension.
+  if ((first_time || new_ids.mid != old_ids.mid ||
+       new_config.rtp.mid != old_config.rtp.mid) &&
+      new_ids.mid != 0 && !new_config.rtp.mid.empty()) {
+    rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpMid::Uri(), new_ids.mid);
+    rtp_rtcp_module_->SetMid(new_config.rtp.mid);
+  }
+
+  if (first_time || new_ids.abs_capture_time != old_ids.abs_capture_time) {
+    absl::string_view uri = AbsoluteCaptureTimeExtension::Uri();
+    rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(uri);
+    if (new_ids.abs_capture_time) {
+      rtp_rtcp_module_->RegisterRtpHeaderExtension(uri,
+                                                   new_ids.abs_capture_time);
+    }
+  }
+
+  if (!ReconfigureSendCodec(new_config)) {
+    RTC_LOG(LS_ERROR) << "Failed to set up send codec state.";
+
+    webrtc::InvokeSetParametersCallback(
+        callback, webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR,
+                                   "Failed to set up send codec state."));
+  }
+
+  // Set currently known overhead (used in ANA, opus only).
+  UpdateOverheadPerPacket();
+
+  channel_send_->CallEncoder([this](AudioEncoder* encoder) {
+    RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+    if (!encoder) {
+      return;
+    }
+    frame_length_range_ = encoder->GetFrameLengthRange();
+    bitrate_range_ = encoder->GetBitrateRange();
+  });
+
+  if (sending_) {
+    ReconfigureBitrateObserver(new_config);
+  }
+
+  config_ = new_config;
+
+  webrtc::InvokeSetParametersCallback(callback, webrtc::RTCError::OK());
+}
+
+void AudioSendStream::Start() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (sending_) {
+    return;
+  }
+  RTC_LOG(LS_INFO) << "AudioSendStream::Start: " << config_.rtp.ssrc;
+  if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
+      config_.max_bitrate_bps != -1 &&
+      (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
+    rtp_transport_->AccountForAudioPacketsInPacedSender(true);
+    rtp_transport_->IncludeOverheadInPacedSender();
+    rtp_rtcp_module_->SetAsPartOfAllocation(true);
+    ConfigureBitrateObserver();
+  } else {
+    rtp_rtcp_module_->SetAsPartOfAllocation(false);
+  }
+  channel_send_->StartSend();
+  sending_ = true;
+  audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
+                                  encoder_num_channels_);
+}
+
+void AudioSendStream::Stop() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!sending_) {
+    return;
+  }
+  RTC_LOG(LS_INFO) << "AudioSendStream::Stop: " << config_.rtp.ssrc;
+  RemoveBitrateObserver();
+  channel_send_->StopSend();
+  sending_ = false;
+  audio_state()->RemoveSendingStream(this);
+}
+
+void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
+  RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
+  RTC_DCHECK_GT(audio_frame->sample_rate_hz_, 0);
+  TRACE_EVENT0("webrtc", "AudioSendStream::SendAudioData");
+  double duration = static_cast<double>(audio_frame->samples_per_channel_) /
+                    audio_frame->sample_rate_hz_;
+  {
+    // Note: SendAudioData() passes the frame further down the pipeline and it
+    // may eventually get sent. But this method is invoked even if we are not
+    // connected, as long as we have an AudioSendStream (created as a result of
+    // an O/A exchange). This means that we are calculating audio levels whether
+    // or not we are sending samples.
+    // TODO(https://crbug.com/webrtc/10771): All "media-source" related stats
+    // should move from send-streams to the local audio sources or tracks; a
+    // send-stream should not be required to read the microphone audio levels.
+    MutexLock lock(&audio_level_lock_);
+    audio_level_.ComputeLevel(*audio_frame, duration);
+  }
+  channel_send_->ProcessAndEncodeAudio(std::move(audio_frame));
+}
+
+bool AudioSendStream::SendTelephoneEvent(int payload_type,
+                                         int payload_frequency,
+                                         int event,
+                                         int duration_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_send_->SetSendTelephoneEventPayloadType(payload_type,
+                                                  payload_frequency);
+  return channel_send_->SendTelephoneEventOutband(event, duration_ms);
+}
+
+void AudioSendStream::SetMuted(bool muted) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_send_->SetInputMute(muted);
+}
+
+webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
+  return GetStats(true);
+}
+
+webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
+    bool has_remote_tracks) const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  webrtc::AudioSendStream::Stats stats;
+  stats.local_ssrc = config_.rtp.ssrc;
+  stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
+
+  webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
+  stats.payload_bytes_sent = call_stats.payload_bytes_sent;
+  stats.header_and_padding_bytes_sent =
+      call_stats.header_and_padding_bytes_sent;
+  stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
+  stats.packets_sent = call_stats.packetsSent;
+  stats.total_packet_send_delay = call_stats.total_packet_send_delay;
+  stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
+  // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
+  // returns 0 to indicate an error value.
+  if (call_stats.rttMs > 0) {
+    stats.rtt_ms = call_stats.rttMs;
+  }
+  if (config_.send_codec_spec) {
+    const auto& spec = *config_.send_codec_spec;
+    stats.codec_name = spec.format.name;
+    stats.codec_payload_type = spec.payload_type;
+
+    // Get data from the last remote RTCP report.
+    for (const ReportBlockData& block :
+         channel_send_->GetRemoteRTCPReportBlocks()) {
+      // Lookup report for send ssrc only.
+      if (block.source_ssrc() == stats.local_ssrc) {
+        stats.packets_lost = block.cumulative_lost();
+        stats.fraction_lost = block.fraction_lost();
+        if (spec.format.clockrate_hz > 0) {
+          stats.jitter_ms = block.jitter(spec.format.clockrate_hz).ms();
+        }
+        break;
+      }
+    }
+  }
+
+  {
+    MutexLock lock(&audio_level_lock_);
+    stats.audio_level = audio_level_.LevelFullRange();
+    stats.total_input_energy = audio_level_.TotalEnergy();
+    stats.total_input_duration = audio_level_.TotalDuration();
+  }
+
+  stats.ana_statistics = channel_send_->GetANAStatistics();
+
+  AudioProcessing* ap = audio_state_->audio_processing();
+  if (ap) {
+    stats.apm_statistics = ap->GetStatistics(has_remote_tracks);
+  }
+
+  stats.report_block_datas = std::move(call_stats.report_block_datas);
+
+  stats.nacks_received = call_stats.nacks_received;
+
+  return stats;
+}
+
+void AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_send_->ReceivedRTCPPacket(packet, length);
+  // Poll if overhead has changed, which it can do if ack triggers us to stop
+  // sending mid/rid.
+  UpdateOverheadPerPacket();
+}
+
+uint32_t AudioSendStream::OnBitrateUpdated(BitrateAllocationUpdate update) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Pick a target bitrate between the constraints. Overrules the allocator if
+  // it 1) allocated a bitrate of zero to disable the stream or 2) allocated a
+  // higher than max to allow for e.g. extra FEC.
+  absl::optional<TargetAudioBitrateConstraints> constraints =
+      GetMinMaxBitrateConstraints();
+  if (constraints) {
+    update.target_bitrate.Clamp(constraints->min, constraints->max);
+    update.stable_target_bitrate.Clamp(constraints->min, constraints->max);
+  }
+  channel_send_->OnBitrateAllocation(update);
+  // The amount of audio protection is not exposed by the encoder, hence
+  // always returning 0.
+  return 0;
+}
+
+void AudioSendStream::SetTransportOverhead(
+    int transport_overhead_per_packet_bytes) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  transport_overhead_per_packet_bytes_ = transport_overhead_per_packet_bytes;
+  UpdateOverheadPerPacket();
+}
+
+void AudioSendStream::UpdateOverheadPerPacket() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  size_t overhead_per_packet_bytes =
+      transport_overhead_per_packet_bytes_ +
+      rtp_rtcp_module_->ExpectedPerPacketOverhead();
+  if (overhead_per_packet_ == overhead_per_packet_bytes) {
+    return;
+  }
+  overhead_per_packet_ = overhead_per_packet_bytes;
+  channel_send_->CallEncoder([&](AudioEncoder* encoder) {
+    encoder->OnReceivedOverhead(overhead_per_packet_bytes);
+  });
+  if (registered_with_allocator_) {
+    ConfigureBitrateObserver();
+  }
+}
+
+size_t AudioSendStream::TestOnlyGetPerPacketOverheadBytes() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return overhead_per_packet_;
+}
+
+RtpState AudioSendStream::GetRtpState() const {
+  return rtp_rtcp_module_->GetRtpState();
+}
+
+const voe::ChannelSendInterface* AudioSendStream::GetChannel() const {
+  return channel_send_.get();
+}
+
+internal::AudioState* AudioSendStream::audio_state() {
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+
+const internal::AudioState* AudioSendStream::audio_state() const {
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+
+void AudioSendStream::StoreEncoderProperties(int sample_rate_hz,
+                                             size_t num_channels) {
+  encoder_sample_rate_hz_ = sample_rate_hz;
+  encoder_num_channels_ = num_channels;
+  if (sending_) {
+    // Update AudioState's information about the stream.
+    audio_state()->AddSendingStream(this, sample_rate_hz, num_channels);
+  }
+}
+
+// Apply current codec settings to a single voe::Channel used for sending.
+bool AudioSendStream::SetupSendCodec(const Config& new_config) {
+  RTC_DCHECK(new_config.send_codec_spec);
+  const auto& spec = *new_config.send_codec_spec;
+
+  RTC_DCHECK(new_config.encoder_factory);
+  std::unique_ptr<AudioEncoder> encoder =
+      new_config.encoder_factory->MakeAudioEncoder(
+          spec.payload_type, spec.format, new_config.codec_pair_id);
+
+  if (!encoder) {
+    RTC_DLOG(LS_ERROR) << "Unable to create encoder for "
+                       << rtc::ToString(spec.format);
+    return false;
+  }
+
+  // If a bitrate has been specified for the codec, use it over the
+  // codec's default.
+  if (spec.target_bitrate_bps) {
+    encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps);
+  }
+
+  // Enable ANA if configured (currently only used by Opus).
+  if (new_config.audio_network_adaptor_config) {
+    if (encoder->EnableAudioNetworkAdaptor(
+            *new_config.audio_network_adaptor_config, event_log_)) {
+      RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
+                       << new_config.rtp.ssrc;
+    } else {
+      RTC_LOG(LS_INFO) << "Failed to enable Audio network adaptor on SSRC "
+                       << new_config.rtp.ssrc;
+    }
+  }
+
+  // Wrap the encoder in an AudioEncoderCNG, if VAD is enabled.
+  if (spec.cng_payload_type) {
+    AudioEncoderCngConfig cng_config;
+    cng_config.num_channels = encoder->NumChannels();
+    cng_config.payload_type = *spec.cng_payload_type;
+    cng_config.speech_encoder = std::move(encoder);
+    cng_config.vad_mode = Vad::kVadNormal;
+    encoder = CreateComfortNoiseEncoder(std::move(cng_config));
+
+    RegisterCngPayloadType(*spec.cng_payload_type,
+                           new_config.send_codec_spec->format.clockrate_hz);
+  }
+
+  // Wrap the encoder in a RED encoder, if RED is enabled.
+  SdpAudioFormat format = spec.format;
+  if (spec.red_payload_type) {
+    AudioEncoderCopyRed::Config red_config;
+    red_config.payload_type = *spec.red_payload_type;
+    red_config.speech_encoder = std::move(encoder);
+    encoder = std::make_unique<AudioEncoderCopyRed>(std::move(red_config),
+                                                    field_trials_);
+    format.name = cricket::kRedCodecName;
+  }
+
+  // Set currently known overhead (used in ANA, opus only).
+  // If overhead changes later, it will be updated in UpdateOverheadPerPacket.
+  if (overhead_per_packet_ > 0) {
+    encoder->OnReceivedOverhead(overhead_per_packet_);
+  }
+
+  StoreEncoderProperties(encoder->SampleRateHz(), encoder->NumChannels());
+  channel_send_->SetEncoder(new_config.send_codec_spec->payload_type, format,
+                            std::move(encoder));
+
+  return true;
+}
+
+bool AudioSendStream::ReconfigureSendCodec(const Config& new_config) {
+  const auto& old_config = config_;
+
+  if (!new_config.send_codec_spec) {
+    // We cannot de-configure a send codec. So we will do nothing.
+    // By design, the send codec should have not been configured.
+    RTC_DCHECK(!old_config.send_codec_spec);
+    return true;
+  }
+
+  if (new_config.send_codec_spec == old_config.send_codec_spec &&
+      new_config.audio_network_adaptor_config ==
+          old_config.audio_network_adaptor_config) {
+    return true;
+  }
+
+  // If we have no encoder, or the format or payload type's changed, create a
+  // new encoder.
+  if (!old_config.send_codec_spec ||
+      new_config.send_codec_spec->format !=
+          old_config.send_codec_spec->format ||
+      new_config.send_codec_spec->payload_type !=
+          old_config.send_codec_spec->payload_type ||
+      new_config.send_codec_spec->red_payload_type !=
+          old_config.send_codec_spec->red_payload_type) {
+    return SetupSendCodec(new_config);
+  }
+
+  const absl::optional<int>& new_target_bitrate_bps =
+      new_config.send_codec_spec->target_bitrate_bps;
+  // If a bitrate has been specified for the codec, use it over the
+  // codec's default.
+  if (new_target_bitrate_bps &&
+      new_target_bitrate_bps !=
+          old_config.send_codec_spec->target_bitrate_bps) {
+    channel_send_->CallEncoder([&](AudioEncoder* encoder) {
+      encoder->OnReceivedTargetAudioBitrate(*new_target_bitrate_bps);
+    });
+  }
+
+  ReconfigureANA(new_config);
+  ReconfigureCNG(new_config);
+
+  return true;
+}
+
+void AudioSendStream::ReconfigureANA(const Config& new_config) {
+  if (new_config.audio_network_adaptor_config ==
+      config_.audio_network_adaptor_config) {
+    return;
+  }
+  if (new_config.audio_network_adaptor_config) {
+    channel_send_->CallEncoder([&](AudioEncoder* encoder) {
+      RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+      if (encoder->EnableAudioNetworkAdaptor(
+              *new_config.audio_network_adaptor_config, event_log_)) {
+        RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
+                         << new_config.rtp.ssrc;
+        if (overhead_per_packet_ > 0) {
+          encoder->OnReceivedOverhead(overhead_per_packet_);
+        }
+      } else {
+        RTC_LOG(LS_INFO) << "Failed to enable Audio network adaptor on SSRC "
+                         << new_config.rtp.ssrc;
+      }
+    });
+  } else {
+    channel_send_->CallEncoder(
+        [&](AudioEncoder* encoder) { encoder->DisableAudioNetworkAdaptor(); });
+    RTC_LOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
+                     << new_config.rtp.ssrc;
+  }
+}
+
+void AudioSendStream::ReconfigureCNG(const Config& new_config) {
+  if (new_config.send_codec_spec->cng_payload_type ==
+      config_.send_codec_spec->cng_payload_type) {
+    return;
+  }
+
+  // Register the CNG payload type if it's been added, don't do anything if CNG
+  // is removed. Payload types must not be redefined.
+  if (new_config.send_codec_spec->cng_payload_type) {
+    RegisterCngPayloadType(*new_config.send_codec_spec->cng_payload_type,
+                           new_config.send_codec_spec->format.clockrate_hz);
+  }
+
+  // Wrap or unwrap the encoder in an AudioEncoderCNG.
+  channel_send_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+    std::unique_ptr<AudioEncoder> old_encoder(std::move(*encoder_ptr));
+    auto sub_encoders = old_encoder->ReclaimContainedEncoders();
+    if (!sub_encoders.empty()) {
+      // Replace enc with its sub encoder. We need to put the sub
+      // encoder in a temporary first, since otherwise the old value
+      // of enc would be destroyed before the new value got assigned,
+      // which would be bad since the new value is a part of the old
+      // value.
+      auto tmp = std::move(sub_encoders[0]);
+      old_encoder = std::move(tmp);
+    }
+    if (new_config.send_codec_spec->cng_payload_type) {
+      AudioEncoderCngConfig config;
+      config.speech_encoder = std::move(old_encoder);
+      config.num_channels = config.speech_encoder->NumChannels();
+      config.payload_type = *new_config.send_codec_spec->cng_payload_type;
+      config.vad_mode = Vad::kVadNormal;
+      *encoder_ptr = CreateComfortNoiseEncoder(std::move(config));
+    } else {
+      *encoder_ptr = std::move(old_encoder);
+    }
+  });
+}
+
+void AudioSendStream::ReconfigureBitrateObserver(
+    const webrtc::AudioSendStream::Config& new_config) {
+  // Since the Config's default is for both of these to be -1, this test will
+  // allow us to configure the bitrate observer if the new config has bitrate
+  // limits set, but would only have us call RemoveBitrateObserver if we were
+  // previously configured with bitrate limits.
+  if (config_.min_bitrate_bps == new_config.min_bitrate_bps &&
+      config_.max_bitrate_bps == new_config.max_bitrate_bps &&
+      config_.bitrate_priority == new_config.bitrate_priority &&
+      TransportSeqNumId(config_) == TransportSeqNumId(new_config) &&
+      config_.audio_network_adaptor_config ==
+          new_config.audio_network_adaptor_config) {
+    return;
+  }
+
+  if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
+      new_config.max_bitrate_bps != -1 && TransportSeqNumId(new_config) != 0) {
+    rtp_transport_->AccountForAudioPacketsInPacedSender(true);
+    rtp_transport_->IncludeOverheadInPacedSender();
+    // We may get a callback immediately as the observer is registered, so
+    // make sure the bitrate limits in config_ are up-to-date.
+    config_.min_bitrate_bps = new_config.min_bitrate_bps;
+    config_.max_bitrate_bps = new_config.max_bitrate_bps;
+
+    config_.bitrate_priority = new_config.bitrate_priority;
+    ConfigureBitrateObserver();
+    rtp_rtcp_module_->SetAsPartOfAllocation(true);
+  } else {
+    rtp_transport_->AccountForAudioPacketsInPacedSender(false);
+    RemoveBitrateObserver();
+    rtp_rtcp_module_->SetAsPartOfAllocation(false);
+  }
+}
+
+void AudioSendStream::ConfigureBitrateObserver() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // This either updates the current observer or adds a new observer.
+  // TODO(srte): Add overhead compensation here.
+  auto constraints = GetMinMaxBitrateConstraints();
+  RTC_DCHECK(constraints.has_value());
+
+  DataRate priority_bitrate = allocation_settings_.priority_bitrate;
+  if (use_legacy_overhead_calculation_) {
+    // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+    constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
+    const TimeDelta kMinPacketDuration = TimeDelta::Millis(20);
+    DataRate max_overhead =
+        DataSize::Bytes(kOverheadPerPacket) / kMinPacketDuration;
+    priority_bitrate += max_overhead;
+  } else {
+    RTC_DCHECK(frame_length_range_);
+    const DataSize overhead_per_packet = DataSize::Bytes(overhead_per_packet_);
+    DataRate min_overhead = overhead_per_packet / frame_length_range_->second;
+    priority_bitrate += min_overhead;
+  }
+
+  if (allocation_settings_.priority_bitrate_raw) {
+    priority_bitrate = *allocation_settings_.priority_bitrate_raw;
+  }
+
+  if (!enable_priority_bitrate_) {
+    priority_bitrate = DataRate::BitsPerSec(0);
+  }
+
+  bitrate_allocator_->AddObserver(
+      this,
+      MediaStreamAllocationConfig{
+          constraints->min.bps<uint32_t>(), constraints->max.bps<uint32_t>(), 0,
+          priority_bitrate.bps(), true,
+          allocation_settings_.bitrate_priority.value_or(
+              config_.bitrate_priority)});
+
+  registered_with_allocator_ = true;
+}
+
+void AudioSendStream::RemoveBitrateObserver() {
+  registered_with_allocator_ = false;
+  bitrate_allocator_->RemoveObserver(this);
+}
+
+absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
+AudioSendStream::GetMinMaxBitrateConstraints() const {
+  if (config_.min_bitrate_bps < 0 || config_.max_bitrate_bps < 0) {
+    RTC_LOG(LS_WARNING) << "Config is invalid: min_bitrate_bps="
+                        << config_.min_bitrate_bps
+                        << "; max_bitrate_bps=" << config_.max_bitrate_bps
+                        << "; both expected greater or equal to 0";
+    return absl::nullopt;
+  }
+  TargetAudioBitrateConstraints constraints{
+      DataRate::BitsPerSec(config_.min_bitrate_bps),
+      DataRate::BitsPerSec(config_.max_bitrate_bps)};
+
+  // If bitrates were explicitly overriden via field trial, use those values.
+  if (allocation_settings_.min_bitrate)
+    constraints.min = *allocation_settings_.min_bitrate;
+  if (allocation_settings_.max_bitrate)
+    constraints.max = *allocation_settings_.max_bitrate;
+
+  // Use encoder defined bitrate range if available.
+  if (bitrate_range_) {
+    constraints.min = bitrate_range_->first;
+    constraints.max = bitrate_range_->second;
+  }
+
+  RTC_DCHECK_GE(constraints.min, DataRate::Zero());
+  RTC_DCHECK_GE(constraints.max, DataRate::Zero());
+  if (constraints.max < constraints.min) {
+    RTC_LOG(LS_WARNING) << "TargetAudioBitrateConstraints::max is less than "
+                        << "TargetAudioBitrateConstraints::min";
+    return absl::nullopt;
+  }
+  if (use_legacy_overhead_calculation_) {
+    // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+    const DataSize kOverheadPerPacket = DataSize::Bytes(20 + 8 + 10 + 12);
+    const TimeDelta kMaxFrameLength =
+        TimeDelta::Millis(60);  // Based on Opus spec
+    const DataRate kMinOverhead = kOverheadPerPacket / kMaxFrameLength;
+    constraints.min += kMinOverhead;
+    constraints.max += kMinOverhead;
+  } else {
+    if (!frame_length_range_.has_value()) {
+      RTC_LOG(LS_WARNING) << "frame_length_range_ is not set";
+      return absl::nullopt;
+    }
+    const DataSize overhead_per_packet = DataSize::Bytes(overhead_per_packet_);
+    constraints.min += overhead_per_packet / frame_length_range_->second;
+    constraints.max += overhead_per_packet / frame_length_range_->first;
+  }
+  return constraints;
+}
+
+void AudioSendStream::RegisterCngPayloadType(int payload_type,
+                                             int clockrate_hz) {
+  channel_send_->RegisterCngPayloadType(payload_type, clockrate_hz);
+}
+
+}  // namespace internal
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/audio_send_stream.h

@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_SEND_STREAM_H_
+#define AUDIO_AUDIO_SEND_STREAM_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "audio/audio_level.h"
+#include "audio/channel_send.h"
+#include "call/audio_send_stream.h"
+#include "call/audio_state.h"
+#include "call/bitrate_allocator.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+class RtcEventLog;
+class RtcpRttStats;
+class RtpTransportControllerSendInterface;
+
+struct AudioAllocationConfig {
+  static constexpr char kKey[] = "WebRTC-Audio-Allocation";
+  // Field Trial configured bitrates to use as overrides over default/user
+  // configured bitrate range when audio bitrate allocation is enabled.
+  absl::optional<DataRate> min_bitrate;
+  absl::optional<DataRate> max_bitrate;
+  DataRate priority_bitrate = DataRate::Zero();
+  // By default the priority_bitrate is compensated for packet overhead.
+  // Use this flag to configure a raw value instead.
+  absl::optional<DataRate> priority_bitrate_raw;
+  absl::optional<double> bitrate_priority;
+
+  std::unique_ptr<StructParametersParser> Parser();
+  explicit AudioAllocationConfig(const FieldTrialsView& field_trials);
+};
+namespace internal {
+class AudioState;
+
+class AudioSendStream final : public webrtc::AudioSendStream,
+                              public webrtc::BitrateAllocatorObserver {
+ public:
+  AudioSendStream(Clock* clock,
+                  const webrtc::AudioSendStream::Config& config,
+                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                  TaskQueueFactory* task_queue_factory,
+                  RtpTransportControllerSendInterface* rtp_transport,
+                  BitrateAllocatorInterface* bitrate_allocator,
+                  RtcEventLog* event_log,
+                  RtcpRttStats* rtcp_rtt_stats,
+                  const absl::optional<RtpState>& suspended_rtp_state,
+                  const FieldTrialsView& field_trials);
+  // For unit tests, which need to supply a mock ChannelSend.
+  AudioSendStream(Clock* clock,
+                  const webrtc::AudioSendStream::Config& config,
+                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                  TaskQueueFactory* task_queue_factory,
+                  RtpTransportControllerSendInterface* rtp_transport,
+                  BitrateAllocatorInterface* bitrate_allocator,
+                  RtcEventLog* event_log,
+                  const absl::optional<RtpState>& suspended_rtp_state,
+                  std::unique_ptr<voe::ChannelSendInterface> channel_send,
+                  const FieldTrialsView& field_trials);
+
+  AudioSendStream() = delete;
+  AudioSendStream(const AudioSendStream&) = delete;
+  AudioSendStream& operator=(const AudioSendStream&) = delete;
+
+  ~AudioSendStream() override;
+
+  // webrtc::AudioSendStream implementation.
+  const webrtc::AudioSendStream::Config& GetConfig() const override;
+  void Reconfigure(const webrtc::AudioSendStream::Config& config,
+                   SetParametersCallback callback) override;
+  void Start() override;
+  void Stop() override;
+  void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
+  bool SendTelephoneEvent(int payload_type,
+                          int payload_frequency,
+                          int event,
+                          int duration_ms) override;
+  void SetMuted(bool muted) override;
+  webrtc::AudioSendStream::Stats GetStats() const override;
+  webrtc::AudioSendStream::Stats GetStats(
+      bool has_remote_tracks) const override;
+
+  void DeliverRtcp(const uint8_t* packet, size_t length);
+
+  // Implements BitrateAllocatorObserver.
+  uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
+
+  void SetTransportOverhead(int transport_overhead_per_packet_bytes);
+
+  RtpState GetRtpState() const;
+  const voe::ChannelSendInterface* GetChannel() const;
+
+  // Returns combined per-packet overhead.
+  size_t TestOnlyGetPerPacketOverheadBytes() const;
+
+ private:
+  class TimedTransport;
+  // Constraints including overhead.
+  struct TargetAudioBitrateConstraints {
+    DataRate min;
+    DataRate max;
+  };
+
+  internal::AudioState* audio_state();
+  const internal::AudioState* audio_state() const;
+
+  void StoreEncoderProperties(int sample_rate_hz, size_t num_channels)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  void ConfigureStream(const Config& new_config,
+                       bool first_time,
+                       SetParametersCallback callback)
+      RTC_RUN_ON(worker_thread_checker_);
+  bool SetupSendCodec(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  bool ReconfigureSendCodec(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  void ReconfigureANA(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  void ReconfigureCNG(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  void ReconfigureBitrateObserver(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  void ConfigureBitrateObserver() RTC_RUN_ON(worker_thread_checker_);
+  void RemoveBitrateObserver() RTC_RUN_ON(worker_thread_checker_);
+
+  // Returns bitrate constraints, maybe including overhead when enabled by
+  // field trial.
+  absl::optional<TargetAudioBitrateConstraints> GetMinMaxBitrateConstraints()
+      const RTC_RUN_ON(worker_thread_checker_);
+
+  // Sets per-packet overhead on encoded (for ANA) based on current known values
+  // of transport and packetization overheads.
+  void UpdateOverheadPerPacket();
+
+  void RegisterCngPayloadType(int payload_type, int clockrate_hz)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  Clock* clock_;
+  const FieldTrialsView& field_trials_;
+
+  SequenceChecker worker_thread_checker_;
+  rtc::RaceChecker audio_capture_race_checker_;
+
+  const bool allocate_audio_without_feedback_;
+  const bool force_no_audio_feedback_ = allocate_audio_without_feedback_;
+  const bool enable_audio_alr_probing_;
+  const AudioAllocationConfig allocation_settings_;
+
+  webrtc::AudioSendStream::Config config_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  const std::unique_ptr<voe::ChannelSendInterface> channel_send_;
+  RtcEventLog* const event_log_;
+  const bool use_legacy_overhead_calculation_;
+  const bool enable_priority_bitrate_;
+
+  int encoder_sample_rate_hz_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
+  size_t encoder_num_channels_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
+  bool sending_ RTC_GUARDED_BY(worker_thread_checker_) = false;
+  mutable Mutex audio_level_lock_;
+  // Keeps track of audio level, total audio energy and total samples duration.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcaudiohandlerstats-totalaudioenergy
+  webrtc::voe::AudioLevel audio_level_ RTC_GUARDED_BY(audio_level_lock_);
+
+  BitrateAllocatorInterface* const bitrate_allocator_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  RtpTransportControllerSendInterface* const rtp_transport_;
+
+  RtpRtcpInterface* const rtp_rtcp_module_;
+  absl::optional<RtpState> const suspended_rtp_state_;
+
+  // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
+  // reserved for padding and MUST NOT be used as a local identifier.
+  // So it should be safe to use 0 here to indicate "not configured".
+  struct ExtensionIds {
+    int audio_level = 0;
+    int abs_send_time = 0;
+    int abs_capture_time = 0;
+    int transport_sequence_number = 0;
+    int mid = 0;
+    int rid = 0;
+    int repaired_rid = 0;
+  };
+  static ExtensionIds FindExtensionIds(
+      const std::vector<RtpExtension>& extensions);
+  static int TransportSeqNumId(const Config& config);
+
+  // Current transport overhead (ICE, TURN, etc.)
+  size_t transport_overhead_per_packet_bytes_
+      RTC_GUARDED_BY(worker_thread_checker_) = 0;
+  // Total overhead, including transport and RTP headers.
+  size_t overhead_per_packet_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
+
+  bool registered_with_allocator_ RTC_GUARDED_BY(worker_thread_checker_) =
+      false;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  absl::optional<std::pair<DataRate, DataRate>> bitrate_range_
+      RTC_GUARDED_BY(worker_thread_checker_);
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_SEND_STREAM_H_

TMessagesProj/jni/voip/webrtc/audio/audio_state.cc

@@ -0,0 +1,212 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_state.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "audio/audio_receive_stream.h"
+#include "audio/audio_send_stream.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace internal {
+
+AudioState::AudioState(const AudioState::Config& config)
+    : config_(config),
+      audio_transport_(config_.audio_mixer.get(),
+                       config_.audio_processing.get(),
+                       config_.async_audio_processing_factory.get()) {
+  RTC_DCHECK(config_.audio_mixer);
+  RTC_DCHECK(config_.audio_device_module);
+}
+
+AudioState::~AudioState() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(receiving_streams_.empty());
+  RTC_DCHECK(sending_streams_.empty());
+  RTC_DCHECK(!null_audio_poller_.Running());
+}
+
+AudioProcessing* AudioState::audio_processing() {
+  return config_.audio_processing.get();
+}
+
+AudioTransport* AudioState::audio_transport() {
+  return &audio_transport_;
+}
+
+void AudioState::AddReceivingStream(
+    webrtc::AudioReceiveStreamInterface* stream) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_EQ(0, receiving_streams_.count(stream));
+  receiving_streams_.insert(stream);
+  if (!config_.audio_mixer->AddSource(
+          static_cast<AudioReceiveStreamImpl*>(stream))) {
+    RTC_DLOG(LS_ERROR) << "Failed to add source to mixer.";
+  }
+
+  // Make sure playback is initialized; start playing if enabled.
+  UpdateNullAudioPollerState();
+  auto* adm = config_.audio_device_module.get();
+  if (!adm->Playing()) {
+    if (adm->InitPlayout() == 0) {
+      if (playout_enabled_) {
+        adm->StartPlayout();
+      }
+    } else {
+      RTC_DLOG_F(LS_ERROR) << "Failed to initialize playout.";
+    }
+  }
+}
+
+void AudioState::RemoveReceivingStream(
+    webrtc::AudioReceiveStreamInterface* stream) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  auto count = receiving_streams_.erase(stream);
+  RTC_DCHECK_EQ(1, count);
+  config_.audio_mixer->RemoveSource(
+      static_cast<AudioReceiveStreamImpl*>(stream));
+  UpdateNullAudioPollerState();
+  if (receiving_streams_.empty()) {
+    config_.audio_device_module->StopPlayout();
+  }
+}
+
+void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
+                                  int sample_rate_hz,
+                                  size_t num_channels) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  auto& properties = sending_streams_[stream];
+  properties.sample_rate_hz = sample_rate_hz;
+  properties.num_channels = num_channels;
+  UpdateAudioTransportWithSendingStreams();
+
+  // Make sure recording is initialized; start recording if enabled.
+  auto* adm = config_.audio_device_module.get();
+  if (!adm->Recording()) {
+    if (adm->InitRecording() == 0) {
+      if (recording_enabled_) {
+        adm->StartRecording();
+      }
+    } else {
+      RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording.";
+    }
+  }
+}
+
+void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  auto count = sending_streams_.erase(stream);
+  RTC_DCHECK_EQ(1, count);
+  UpdateAudioTransportWithSendingStreams();
+  if (sending_streams_.empty()) {
+    config_.audio_device_module->StopRecording();
+  }
+}
+
+void AudioState::SetPlayout(bool enabled) {
+  RTC_LOG(LS_INFO) << "SetPlayout(" << enabled << ")";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (playout_enabled_ != enabled) {
+    playout_enabled_ = enabled;
+    if (enabled) {
+      UpdateNullAudioPollerState();
+      if (!receiving_streams_.empty()) {
+        config_.audio_device_module->StartPlayout();
+      }
+    } else {
+      config_.audio_device_module->StopPlayout();
+      UpdateNullAudioPollerState();
+    }
+  }
+}
+
+void AudioState::SetRecording(bool enabled) {
+  RTC_LOG(LS_INFO) << "SetRecording(" << enabled << ")";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (recording_enabled_ != enabled) {
+    recording_enabled_ = enabled;
+    if (enabled) {
+      if (!sending_streams_.empty()) {
+        config_.audio_device_module->StartRecording();
+      }
+    } else {
+      config_.audio_device_module->StopRecording();
+    }
+  }
+}
+
+void AudioState::SetStereoChannelSwapping(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_transport_.SetStereoChannelSwapping(enable);
+}
+
+void AudioState::UpdateAudioTransportWithSendingStreams() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  std::vector<AudioSender*> audio_senders;
+  int max_sample_rate_hz = 8000;
+  size_t max_num_channels = 1;
+  for (const auto& kv : sending_streams_) {
+    audio_senders.push_back(kv.first);
+    max_sample_rate_hz = std::max(max_sample_rate_hz, kv.second.sample_rate_hz);
+    max_num_channels = std::max(max_num_channels, kv.second.num_channels);
+  }
+  audio_transport_.UpdateAudioSenders(std::move(audio_senders),
+                                      max_sample_rate_hz, max_num_channels);
+}
+
+void AudioState::UpdateNullAudioPollerState() {
+  // Run NullAudioPoller when there are receiving streams and playout is
+  // disabled.
+  if (!receiving_streams_.empty() && !playout_enabled_) {
+    if (!null_audio_poller_.Running()) {
+      AudioTransport* audio_transport = &audio_transport_;
+      null_audio_poller_ = RepeatingTaskHandle::Start(
+          TaskQueueBase::Current(), [audio_transport] {
+            static constexpr size_t kNumChannels = 1;
+            static constexpr uint32_t kSamplesPerSecond = 48'000;
+            // 10ms of samples
+            static constexpr size_t kNumSamples = kSamplesPerSecond / 100;
+
+            // Buffer to hold the audio samples.
+            int16_t buffer[kNumSamples * kNumChannels];
+
+            // Output variables from `NeedMorePlayData`.
+            size_t n_samples;
+            int64_t elapsed_time_ms;
+            int64_t ntp_time_ms;
+            audio_transport->NeedMorePlayData(
+                kNumSamples, sizeof(int16_t), kNumChannels, kSamplesPerSecond,
+                buffer, n_samples, &elapsed_time_ms, &ntp_time_ms);
+
+            // Reschedule the next poll iteration.
+            return TimeDelta::Millis(10);
+          });
+    }
+  } else {
+    null_audio_poller_.Stop();
+  }
+}
+}  // namespace internal
+
+rtc::scoped_refptr<AudioState> AudioState::Create(
+    const AudioState::Config& config) {
+  return rtc::make_ref_counted<internal::AudioState>(config);
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/audio_state.h

@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_STATE_H_
+#define AUDIO_AUDIO_STATE_H_
+
+#include <map>
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "audio/audio_transport_impl.h"
+#include "call/audio_state.h"
+#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioSendStream;
+class AudioReceiveStreamInterface;
+
+namespace internal {
+
+class AudioState : public webrtc::AudioState {
+ public:
+  explicit AudioState(const AudioState::Config& config);
+
+  AudioState() = delete;
+  AudioState(const AudioState&) = delete;
+  AudioState& operator=(const AudioState&) = delete;
+
+  ~AudioState() override;
+
+  AudioProcessing* audio_processing() override;
+  AudioTransport* audio_transport() override;
+
+  void SetPlayout(bool enabled) override;
+  void SetRecording(bool enabled) override;
+
+  void SetStereoChannelSwapping(bool enable) override;
+
+  AudioDeviceModule* audio_device_module() {
+    RTC_DCHECK(config_.audio_device_module);
+    return config_.audio_device_module.get();
+  }
+
+  void AddReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
+  void RemoveReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
+
+  void AddSendingStream(webrtc::AudioSendStream* stream,
+                        int sample_rate_hz,
+                        size_t num_channels);
+  void RemoveSendingStream(webrtc::AudioSendStream* stream);
+
+ private:
+  void UpdateAudioTransportWithSendingStreams();
+  void UpdateNullAudioPollerState() RTC_RUN_ON(&thread_checker_);
+
+  SequenceChecker thread_checker_;
+  SequenceChecker process_thread_checker_{SequenceChecker::kDetached};
+  const webrtc::AudioState::Config config_;
+  bool recording_enabled_ = true;
+  bool playout_enabled_ = true;
+
+  // Transports mixed audio from the mixer to the audio device and
+  // recorded audio to the sending streams.
+  AudioTransportImpl audio_transport_;
+
+  // Null audio poller is used to continue polling the audio streams if audio
+  // playout is disabled so that audio processing still happens and the audio
+  // stats are still updated.
+  RepeatingTaskHandle null_audio_poller_ RTC_GUARDED_BY(&thread_checker_);
+
+  webrtc::flat_set<webrtc::AudioReceiveStreamInterface*> receiving_streams_;
+  struct StreamProperties {
+    int sample_rate_hz = 0;
+    size_t num_channels = 0;
+  };
+  std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_STATE_H_

TMessagesProj/jni/voip/webrtc/audio/audio_transport_impl.cc

@@ -0,0 +1,285 @@
+/*
+ *  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 "audio/audio_transport_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "audio/remix_resample.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/audio_sender.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
+#include "modules/audio_processing/include/audio_frame_proxies.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+
+// We want to process at the lowest sample rate and channel count possible
+// without losing information. Choose the lowest native rate at least equal to
+// the minimum of input and codec rates, choose lowest channel count, and
+// configure the audio frame.
+void InitializeCaptureFrame(int input_sample_rate,
+                            int send_sample_rate_hz,
+                            size_t input_num_channels,
+                            size_t send_num_channels,
+                            AudioFrame* audio_frame) {
+  RTC_DCHECK(audio_frame);
+  int min_processing_rate_hz = std::min(input_sample_rate, send_sample_rate_hz);
+  for (int native_rate_hz : AudioProcessing::kNativeSampleRatesHz) {
+    audio_frame->sample_rate_hz_ = native_rate_hz;
+    if (audio_frame->sample_rate_hz_ >= min_processing_rate_hz) {
+      break;
+    }
+  }
+  audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
+}
+
+void ProcessCaptureFrame(uint32_t delay_ms,
+                         bool key_pressed,
+                         bool swap_stereo_channels,
+                         AudioProcessing* audio_processing,
+                         AudioFrame* audio_frame) {
+  RTC_DCHECK(audio_frame);
+  if (audio_processing) {
+    audio_processing->set_stream_delay_ms(delay_ms);
+    audio_processing->set_stream_key_pressed(key_pressed);
+    int error = ProcessAudioFrame(audio_processing, audio_frame);
+
+    RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
+  }
+
+  if (swap_stereo_channels) {
+    AudioFrameOperations::SwapStereoChannels(audio_frame);
+  }
+}
+
+// Resample audio in `frame` to given sample rate preserving the
+// channel count and place the result in `destination`.
+int Resample(const AudioFrame& frame,
+             const int destination_sample_rate,
+             PushResampler<int16_t>* resampler,
+             int16_t* destination) {
+  TRACE_EVENT2("webrtc", "Resample", "frame sample rate", frame.sample_rate_hz_,
+               "destination_sample_rate", destination_sample_rate);
+  const int number_of_channels = static_cast<int>(frame.num_channels_);
+  const int target_number_of_samples_per_channel =
+      destination_sample_rate / 100;
+  resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
+                                number_of_channels);
+
+  // TODO(yujo): make resampler take an AudioFrame, and add special case
+  // handling of muted frames.
+  return resampler->Resample(
+      frame.data(), frame.samples_per_channel_ * number_of_channels,
+      destination, number_of_channels * target_number_of_samples_per_channel);
+}
+}  // namespace
+
+AudioTransportImpl::AudioTransportImpl(
+    AudioMixer* mixer,
+    AudioProcessing* audio_processing,
+    AsyncAudioProcessing::Factory* async_audio_processing_factory)
+    : audio_processing_(audio_processing),
+      async_audio_processing_(
+          async_audio_processing_factory
+              ? async_audio_processing_factory->CreateAsyncAudioProcessing(
+                    [this](std::unique_ptr<AudioFrame> frame) {
+                      this->SendProcessedData(std::move(frame));
+                    })
+              : nullptr),
+      mixer_(mixer) {
+  RTC_DCHECK(mixer);
+}
+
+AudioTransportImpl::~AudioTransportImpl() {}
+
+int32_t AudioTransportImpl::RecordedDataIsAvailable(
+    const void* audio_data,
+    size_t number_of_frames,
+    size_t bytes_per_sample,
+    size_t number_of_channels,
+    uint32_t sample_rate,
+    uint32_t audio_delay_milliseconds,
+    int32_t clock_drift,
+    uint32_t volume,
+    bool key_pressed,
+    uint32_t& new_mic_volume) {  // NOLINT: to avoid changing APIs
+  return RecordedDataIsAvailable(
+      audio_data, number_of_frames, bytes_per_sample, number_of_channels,
+      sample_rate, audio_delay_milliseconds, clock_drift, volume, key_pressed,
+      new_mic_volume, /*estimated_capture_time_ns=*/absl::nullopt);
+}
+
+// Not used in Chromium. Process captured audio and distribute to all sending
+// streams, and try to do this at the lowest possible sample rate.
+int32_t AudioTransportImpl::RecordedDataIsAvailable(
+    const void* audio_data,
+    size_t number_of_frames,
+    size_t bytes_per_sample,
+    size_t number_of_channels,
+    uint32_t sample_rate,
+    uint32_t audio_delay_milliseconds,
+    int32_t /*clock_drift*/,
+    uint32_t /*volume*/,
+    bool key_pressed,
+    uint32_t& /*new_mic_volume*/,
+    absl::optional<int64_t>
+        estimated_capture_time_ns) {  // NOLINT: to avoid changing APIs
+  RTC_DCHECK(audio_data);
+  RTC_DCHECK_GE(number_of_channels, 1);
+  RTC_DCHECK_LE(number_of_channels, 2);
+  RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
+  RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
+  // 100 = 1 second / data duration (10 ms).
+  RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
+  RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
+                AudioFrame::kMaxDataSizeBytes);
+
+  int send_sample_rate_hz = 0;
+  size_t send_num_channels = 0;
+  bool swap_stereo_channels = false;
+  {
+    MutexLock lock(&capture_lock_);
+    send_sample_rate_hz = send_sample_rate_hz_;
+    send_num_channels = send_num_channels_;
+    swap_stereo_channels = swap_stereo_channels_;
+  }
+
+  std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
+  InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels,
+                         send_num_channels, audio_frame.get());
+  voe::RemixAndResample(static_cast<const int16_t*>(audio_data),
+                        number_of_frames, number_of_channels, sample_rate,
+                        &capture_resampler_, audio_frame.get());
+  ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
+                      swap_stereo_channels, audio_processing_,
+                      audio_frame.get());
+
+  if (estimated_capture_time_ns) {
+    audio_frame->set_absolute_capture_timestamp_ms(*estimated_capture_time_ns /
+                                                   1000000);
+  }
+
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  if (async_audio_processing_)
+    async_audio_processing_->Process(std::move(audio_frame));
+  else
+    SendProcessedData(std::move(audio_frame));
+
+  return 0;
+}
+
+void AudioTransportImpl::SendProcessedData(
+    std::unique_ptr<AudioFrame> audio_frame) {
+  TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  MutexLock lock(&capture_lock_);
+  if (audio_senders_.empty())
+    return;
+
+  auto it = audio_senders_.begin();
+  while (++it != audio_senders_.end()) {
+    auto audio_frame_copy = std::make_unique<AudioFrame>();
+    audio_frame_copy->CopyFrom(*audio_frame);
+    (*it)->SendAudioData(std::move(audio_frame_copy));
+  }
+  // Send the original frame to the first stream w/o copying.
+  (*audio_senders_.begin())->SendAudioData(std::move(audio_frame));
+}
+
+// Mix all received streams, feed the result to the AudioProcessing module, then
+// resample the result to the requested output rate.
+int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
+                                             const size_t nBytesPerSample,
+                                             const size_t nChannels,
+                                             const uint32_t samplesPerSec,
+                                             void* audioSamples,
+                                             size_t& nSamplesOut,
+                                             int64_t* elapsed_time_ms,
+                                             int64_t* ntp_time_ms) {
+  TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
+  RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
+  RTC_DCHECK_GE(nChannels, 1);
+  RTC_DCHECK_LE(nChannels, 2);
+  RTC_DCHECK_GE(
+      samplesPerSec,
+      static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
+
+  // 100 = 1 second / data duration (10 ms).
+  RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
+  RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
+                AudioFrame::kMaxDataSizeBytes);
+
+  mixer_->Mix(nChannels, &mixed_frame_);
+  *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
+  *ntp_time_ms = mixed_frame_.ntp_time_ms_;
+
+  if (audio_processing_) {
+    const auto error =
+        ProcessReverseAudioFrame(audio_processing_, &mixed_frame_);
+    RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
+  }
+
+  nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_,
+                         static_cast<int16_t*>(audioSamples));
+  RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
+  return 0;
+}
+
+// Used by Chromium - same as NeedMorePlayData() but because Chrome has its
+// own APM instance, does not call audio_processing_->ProcessReverseStream().
+void AudioTransportImpl::PullRenderData(int bits_per_sample,
+                                        int sample_rate,
+                                        size_t number_of_channels,
+                                        size_t number_of_frames,
+                                        void* audio_data,
+                                        int64_t* elapsed_time_ms,
+                                        int64_t* ntp_time_ms) {
+  TRACE_EVENT2("webrtc", "AudioTransportImpl::PullRenderData", "sample_rate",
+               sample_rate, "number_of_frames", number_of_frames);
+  RTC_DCHECK_EQ(bits_per_sample, 16);
+  RTC_DCHECK_GE(number_of_channels, 1);
+  RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
+
+  // 100 = 1 second / data duration (10 ms).
+  RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
+
+  // 8 = bits per byte.
+  RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
+                AudioFrame::kMaxDataSizeBytes);
+  mixer_->Mix(number_of_channels, &mixed_frame_);
+  *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
+  *ntp_time_ms = mixed_frame_.ntp_time_ms_;
+
+  auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_,
+                                 static_cast<int16_t*>(audio_data));
+  RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
+}
+
+void AudioTransportImpl::UpdateAudioSenders(std::vector<AudioSender*> senders,
+                                            int send_sample_rate_hz,
+                                            size_t send_num_channels) {
+  MutexLock lock(&capture_lock_);
+  audio_senders_ = std::move(senders);
+  send_sample_rate_hz_ = send_sample_rate_hz;
+  send_num_channels_ = send_num_channels;
+}
+
+void AudioTransportImpl::SetStereoChannelSwapping(bool enable) {
+  MutexLock lock(&capture_lock_);
+  swap_stereo_channels_ = enable;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/audio_transport_impl.h

@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_
+#define AUDIO_AUDIO_TRANSPORT_IMPL_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_mixer.h"
+#include "api/scoped_refptr.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioSender;
+
+class AudioTransportImpl : public AudioTransport {
+ public:
+  AudioTransportImpl(
+      AudioMixer* mixer,
+      AudioProcessing* audio_processing,
+      AsyncAudioProcessing::Factory* async_audio_processing_factory);
+
+  AudioTransportImpl() = delete;
+  AudioTransportImpl(const AudioTransportImpl&) = delete;
+  AudioTransportImpl& operator=(const AudioTransportImpl&) = delete;
+
+  ~AudioTransportImpl() override;
+
+  // TODO(bugs.webrtc.org/13620) Deprecate this function
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  size_t nSamples,
+                                  size_t nBytesPerSample,
+                                  size_t nChannels,
+                                  uint32_t samplesPerSec,
+                                  uint32_t totalDelayMS,
+                                  int32_t clockDrift,
+                                  uint32_t currentMicLevel,
+                                  bool keyPressed,
+                                  uint32_t& newMicLevel) override;
+
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samplesPerSec,
+      uint32_t totalDelayMS,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> estimated_capture_time_ns) override;
+
+  int32_t NeedMorePlayData(size_t nSamples,
+                           size_t nBytesPerSample,
+                           size_t nChannels,
+                           uint32_t samplesPerSec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override;
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override;
+
+  void UpdateAudioSenders(std::vector<AudioSender*> senders,
+                          int send_sample_rate_hz,
+                          size_t send_num_channels);
+  void SetStereoChannelSwapping(bool enable);
+
+ private:
+  void SendProcessedData(std::unique_ptr<AudioFrame> audio_frame);
+
+  // Shared.
+  AudioProcessing* audio_processing_ = nullptr;
+
+  // Capture side.
+
+  // Thread-safe.
+  const std::unique_ptr<AsyncAudioProcessing> async_audio_processing_;
+
+  mutable Mutex capture_lock_;
+  std::vector<AudioSender*> audio_senders_ RTC_GUARDED_BY(capture_lock_);
+  int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
+  size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
+  bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
+  PushResampler<int16_t> capture_resampler_;
+
+  // Render side.
+
+  rtc::scoped_refptr<AudioMixer> mixer_;
+  AudioFrame mixed_frame_;
+  // Converts mixed audio to the audio device output rate.
+  PushResampler<int16_t> render_resampler_;
+};
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_TRANSPORT_IMPL_H_

TMessagesProj/jni/voip/webrtc/audio/channel_receive.h

@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CHANNEL_RECEIVE_H_
+#define AUDIO_CHANNEL_RECEIVE_H_
+
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/call/audio_sink.h"
+#include "api/call/transport.h"
+#include "api/crypto/crypto_options.h"
+#include "api/frame_transformer_interface.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "call/syncable.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/rtp_rtcp/source/source_tracker.h"
+#include "system_wrappers/include/clock.h"
+
+// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
+// warnings about use of unsigned short.
+// These need cleanup, in a separate cl.
+
+namespace rtc {
+class TimestampWrapAroundHandler;
+}
+
+namespace webrtc {
+
+class AudioDeviceModule;
+class FrameDecryptorInterface;
+class PacketRouter;
+class RateLimiter;
+class ReceiveStatistics;
+class RtcEventLog;
+class RtpPacketReceived;
+class RtpRtcp;
+
+struct CallReceiveStatistics {
+  int cumulativeLost;
+  unsigned int jitterSamples;
+  int64_t payload_bytes_received = 0;
+  int64_t header_and_padding_bytes_received = 0;
+  int packetsReceived;
+  uint32_t nacks_sent = 0;
+  // The capture NTP time (in local timebase) of the first played out audio
+  // frame.
+  int64_t capture_start_ntp_time_ms_;
+  // The timestamp at which the last packet was received, i.e. the time of the
+  // local clock when it was received - not the RTP timestamp of that packet.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
+  absl::optional<Timestamp> last_packet_received;
+  // Remote outbound stats derived by the received RTCP sender reports.
+  // Note that the timestamps below correspond to the time elapsed since the
+  // Unix epoch.
+  // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
+  absl::optional<int64_t> last_sender_report_timestamp_ms;
+  absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+  uint64_t sender_reports_packets_sent = 0;
+  uint64_t sender_reports_bytes_sent = 0;
+  uint64_t sender_reports_reports_count = 0;
+  absl::optional<TimeDelta> round_trip_time;
+  TimeDelta total_round_trip_time = TimeDelta::Zero();
+  int round_trip_time_measurements;
+};
+
+namespace voe {
+
+class ChannelSendInterface;
+
+// Interface class needed for AudioReceiveStreamInterface tests that use a
+// MockChannelReceive.
+
+class ChannelReceiveInterface : public RtpPacketSinkInterface {
+ public:
+  virtual ~ChannelReceiveInterface() = default;
+
+  virtual void SetSink(AudioSinkInterface* sink) = 0;
+
+  virtual void SetReceiveCodecs(
+      const std::map<int, SdpAudioFormat>& codecs) = 0;
+
+  virtual void StartPlayout() = 0;
+  virtual void StopPlayout() = 0;
+
+  // Payload type and format of last received RTP packet, if any.
+  virtual absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
+      const = 0;
+
+  virtual void ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0;
+
+  virtual void SetChannelOutputVolumeScaling(float scaling) = 0;
+  virtual int GetSpeechOutputLevelFullRange() const = 0;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec:
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  virtual double GetTotalOutputEnergy() const = 0;
+  virtual double GetTotalOutputDuration() const = 0;
+
+  // Stats.
+  virtual NetworkStatistics GetNetworkStatistics(
+      bool get_and_clear_legacy_stats) const = 0;
+  virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0;
+
+  // Audio+Video Sync.
+  virtual uint32_t GetDelayEstimate() const = 0;
+  virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0;
+  virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                                      int64_t* time_ms) const = 0;
+  virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                                 int64_t time_ms) = 0;
+  virtual absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
+      int64_t now_ms) const = 0;
+
+  // Audio quality.
+  // Base minimum delay sets lower bound on minimum delay value which
+  // determines minimum delay until audio playout.
+  virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
+  virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
+
+  // Produces the transport-related timestamps; current_delay_ms is left unset.
+  virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
+
+  virtual void RegisterReceiverCongestionControlObjects(
+      PacketRouter* packet_router) = 0;
+  virtual void ResetReceiverCongestionControlObjects() = 0;
+
+  virtual CallReceiveStatistics GetRTCPStatistics() const = 0;
+  virtual void SetNACKStatus(bool enable, int max_packets) = 0;
+  virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
+
+  virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sample_rate_hz,
+      AudioFrame* audio_frame) = 0;
+
+  virtual int PreferredSampleRate() const = 0;
+
+  // Sets the source tracker to notify about "delivered" packets when output is
+  // muted.
+  virtual void SetSourceTracker(SourceTracker* source_tracker) = 0;
+
+  // Associate to a send channel.
+  // Used for obtaining RTT for a receive-only channel.
+  virtual void SetAssociatedSendChannel(
+      const ChannelSendInterface* channel) = 0;
+
+  // Sets a frame transformer between the depacketizer and the decoder, to
+  // transform the received frames before decoding them.
+  virtual void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+          frame_transformer) = 0;
+
+  virtual void SetFrameDecryptor(
+      rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
+
+  virtual void OnLocalSsrcChange(uint32_t local_ssrc) = 0;
+  virtual uint32_t GetLocalSsrc() const = 0;
+};
+
+std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
+    Clock* clock,
+    NetEqFactory* neteq_factory,
+    AudioDeviceModule* audio_device_module,
+    Transport* rtcp_send_transport,
+    RtcEventLog* rtc_event_log,
+    uint32_t local_ssrc,
+    uint32_t remote_ssrc,
+    size_t jitter_buffer_max_packets,
+    bool jitter_buffer_fast_playout,
+    int jitter_buffer_min_delay_ms,
+    bool enable_non_sender_rtt,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_CHANNEL_RECEIVE_H_

TMessagesProj/jni/voip/webrtc/audio/channel_receive_frame_transformer_delegate.cc

@@ -0,0 +1,167 @@
+/*
+ *  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 "audio/channel_receive_frame_transformer_delegate.h"
+
+#include <string>
+#include <utility>
+
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace {
+
+class TransformableIncomingAudioFrame
+    : public TransformableAudioFrameInterface {
+ public:
+  TransformableIncomingAudioFrame(rtc::ArrayView<const uint8_t> payload,
+                                  const RTPHeader& header,
+                                  uint32_t ssrc,
+                                  const std::string& codec_mime_type)
+      : payload_(payload.data(), payload.size()),
+        header_(header),
+        ssrc_(ssrc),
+        codec_mime_type_(codec_mime_type) {}
+  ~TransformableIncomingAudioFrame() override = default;
+  rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
+
+  void SetData(rtc::ArrayView<const uint8_t> data) override {
+    payload_.SetData(data.data(), data.size());
+  }
+
+  void SetRTPTimestamp(uint32_t timestamp) override {
+    header_.timestamp = timestamp;
+  }
+
+  uint8_t GetPayloadType() const override { return header_.payloadType; }
+  uint32_t GetSsrc() const override { return ssrc_; }
+  uint32_t GetTimestamp() const override { return header_.timestamp; }
+  rtc::ArrayView<const uint32_t> GetContributingSources() const override {
+    return rtc::ArrayView<const uint32_t>(header_.arrOfCSRCs, header_.numCSRCs);
+  }
+  Direction GetDirection() const override { return Direction::kReceiver; }
+
+  std::string GetMimeType() const override { return codec_mime_type_; }
+  const absl::optional<uint16_t> SequenceNumber() const override {
+    return header_.sequenceNumber;
+  }
+
+  absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
+    // This could be extracted from received header extensions + extrapolation,
+    // if required in future, eg for being able to re-send received frames.
+    return absl::nullopt;
+  }
+  const RTPHeader& Header() const { return header_; }
+
+  FrameType Type() const override {
+    return header_.extension.voiceActivity ? FrameType::kAudioFrameSpeech
+                                           : FrameType::kAudioFrameCN;
+  }
+
+ private:
+  rtc::Buffer payload_;
+  RTPHeader header_;
+  uint32_t ssrc_;
+  std::string codec_mime_type_;
+};
+}  // namespace
+
+ChannelReceiveFrameTransformerDelegate::ChannelReceiveFrameTransformerDelegate(
+    ReceiveFrameCallback receive_frame_callback,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    TaskQueueBase* channel_receive_thread)
+    : receive_frame_callback_(receive_frame_callback),
+      frame_transformer_(std::move(frame_transformer)),
+      channel_receive_thread_(channel_receive_thread) {}
+
+void ChannelReceiveFrameTransformerDelegate::Init() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  frame_transformer_->RegisterTransformedFrameCallback(
+      rtc::scoped_refptr<TransformedFrameCallback>(this));
+}
+
+void ChannelReceiveFrameTransformerDelegate::Reset() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  frame_transformer_->UnregisterTransformedFrameCallback();
+  frame_transformer_ = nullptr;
+  receive_frame_callback_ = ReceiveFrameCallback();
+}
+
+void ChannelReceiveFrameTransformerDelegate::Transform(
+    rtc::ArrayView<const uint8_t> packet,
+    const RTPHeader& header,
+    uint32_t ssrc,
+    const std::string& codec_mime_type) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (short_circuit_) {
+    receive_frame_callback_(packet, header);
+  } else {
+    frame_transformer_->Transform(
+        std::make_unique<TransformableIncomingAudioFrame>(packet, header, ssrc,
+                                                          codec_mime_type));
+  }
+}
+
+void ChannelReceiveFrameTransformerDelegate::OnTransformedFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) {
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate(this);
+  channel_receive_thread_->PostTask(
+      [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
+        delegate->ReceiveFrame(std::move(frame));
+      });
+}
+
+void ChannelReceiveFrameTransformerDelegate::StartShortCircuiting() {
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate(this);
+  channel_receive_thread_->PostTask([delegate = std::move(delegate)]() mutable {
+    RTC_DCHECK_RUN_ON(&delegate->sequence_checker_);
+    delegate->short_circuit_ = true;
+  });
+}
+
+void ChannelReceiveFrameTransformerDelegate::ReceiveFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (!receive_frame_callback_)
+    return;
+
+  RTPHeader header;
+  if (frame->GetDirection() ==
+      TransformableFrameInterface::Direction::kSender) {
+    auto* transformed_frame =
+        static_cast<TransformableAudioFrameInterface*>(frame.get());
+    header.payloadType = transformed_frame->GetPayloadType();
+    header.timestamp = transformed_frame->GetTimestamp();
+    header.ssrc = transformed_frame->GetSsrc();
+    if (transformed_frame->AbsoluteCaptureTimestamp().has_value()) {
+      header.extension.absolute_capture_time = AbsoluteCaptureTime();
+      header.extension.absolute_capture_time->absolute_capture_timestamp =
+          transformed_frame->AbsoluteCaptureTimestamp().value();
+    }
+  } else {
+    auto* transformed_frame =
+        static_cast<TransformableIncomingAudioFrame*>(frame.get());
+    header = transformed_frame->Header();
+  }
+
+  // TODO(crbug.com/1464860): Take an explicit struct with the required
+  // information rather than the RTPHeader to make it easier to
+  // construct the required information when injecting transformed frames not
+  // originally from this receiver.
+  receive_frame_callback_(frame->GetData(), header);
+}
+
+rtc::scoped_refptr<FrameTransformerInterface>
+ChannelReceiveFrameTransformerDelegate::FrameTransformer() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return frame_transformer_;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/channel_receive_frame_transformer_delegate.h

@@ -0,0 +1,81 @@
+/*
+ *  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 AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_
+#define AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_
+
+#include <memory>
+#include <string>
+
+#include "api/frame_transformer_interface.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// Delegates calls to FrameTransformerInterface to transform frames, and to
+// ChannelReceive to receive the transformed frames using the
+// `receive_frame_callback_` on the `channel_receive_thread_`.
+class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback {
+ public:
+  using ReceiveFrameCallback =
+      std::function<void(rtc::ArrayView<const uint8_t> packet,
+                         const RTPHeader& header)>;
+  ChannelReceiveFrameTransformerDelegate(
+      ReceiveFrameCallback receive_frame_callback,
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+      TaskQueueBase* channel_receive_thread);
+
+  // Registers `this` as callback for `frame_transformer_`, to get the
+  // transformed frames.
+  void Init();
+
+  // Unregisters and releases the `frame_transformer_` reference, and resets
+  // `receive_frame_callback_` on `channel_receive_thread_`. Called from
+  // ChannelReceive destructor to prevent running the callback on a dangling
+  // channel.
+  void Reset();
+
+  // Delegates the call to FrameTransformerInterface::Transform, to transform
+  // the frame asynchronously.
+  void Transform(rtc::ArrayView<const uint8_t> packet,
+                 const RTPHeader& header,
+                 uint32_t ssrc,
+                 const std::string& codec_mime_type);
+
+  // Implements TransformedFrameCallback. Can be called on any thread.
+  void OnTransformedFrame(
+      std::unique_ptr<TransformableFrameInterface> frame) override;
+
+  void StartShortCircuiting() override;
+
+  // Delegates the call to ChannelReceive::OnReceivedPayloadData on the
+  // `channel_receive_thread_`, by calling `receive_frame_callback_`.
+  void ReceiveFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
+
+  rtc::scoped_refptr<FrameTransformerInterface> FrameTransformer();
+
+ protected:
+  ~ChannelReceiveFrameTransformerDelegate() override = default;
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  ReceiveFrameCallback receive_frame_callback_
+      RTC_GUARDED_BY(sequence_checker_);
+  rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
+      RTC_GUARDED_BY(sequence_checker_);
+  TaskQueueBase* const channel_receive_thread_;
+  bool short_circuit_ RTC_GUARDED_BY(sequence_checker_) = false;
+};
+
+}  // namespace webrtc
+#endif  // AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_

TMessagesProj/jni/voip/webrtc/audio/channel_send.cc

@@ -0,0 +1,908 @@
+/*
+ *  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 "audio/channel_send.h"
+
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/call/transport.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/sequence_checker.h"
+#include "audio/channel_send_frame_transformer_delegate.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_processing/rms_level.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace voe {
+
+namespace {
+
+constexpr int64_t kMaxRetransmissionWindowMs = 1000;
+constexpr int64_t kMinRetransmissionWindowMs = 30;
+
+class RtpPacketSenderProxy;
+class TransportSequenceNumberProxy;
+
+class ChannelSend : public ChannelSendInterface,
+                    public AudioPacketizationCallback,  // receive encoded
+                                                        // packets from the ACM
+                    public RtcpPacketTypeCounterObserver,
+                    public ReportBlockDataObserver {
+ public:
+  ChannelSend(Clock* clock,
+              TaskQueueFactory* task_queue_factory,
+              Transport* rtp_transport,
+              RtcpRttStats* rtcp_rtt_stats,
+              RtcEventLog* rtc_event_log,
+              FrameEncryptorInterface* frame_encryptor,
+              const webrtc::CryptoOptions& crypto_options,
+              bool extmap_allow_mixed,
+              int rtcp_report_interval_ms,
+              uint32_t ssrc,
+              rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+              RtpTransportControllerSendInterface* transport_controller,
+              const FieldTrialsView& field_trials);
+
+  ~ChannelSend() override;
+
+  // Send using this encoder, with this payload type.
+  void SetEncoder(int payload_type,
+                  const SdpAudioFormat& encoder_format,
+                  std::unique_ptr<AudioEncoder> encoder) override;
+  void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
+                         modifier) override;
+  void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) override;
+
+  // API methods
+  void StartSend() override;
+  void StopSend() override;
+
+  // Codecs
+  void OnBitrateAllocation(BitrateAllocationUpdate update) override;
+  int GetTargetBitrate() const override;
+
+  // Network
+  void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
+
+  // Muting, Volume and Level.
+  void SetInputMute(bool enable) override;
+
+  // Stats.
+  ANAStats GetANAStatistics() const override;
+
+  // Used by AudioSendStream.
+  RtpRtcpInterface* GetRtpRtcp() const override;
+
+  void RegisterCngPayloadType(int payload_type, int payload_frequency) override;
+
+  // DTMF.
+  bool SendTelephoneEventOutband(int event, int duration_ms) override;
+  void SetSendTelephoneEventPayloadType(int payload_type,
+                                        int payload_frequency) override;
+
+  // RTP+RTCP
+  void SetSendAudioLevelIndicationStatus(bool enable, int id) override;
+
+  void RegisterSenderCongestionControlObjects(
+      RtpTransportControllerSendInterface* transport) override;
+  void ResetSenderCongestionControlObjects() override;
+  void SetRTCP_CNAME(absl::string_view c_name) override;
+  std::vector<ReportBlockData> GetRemoteRTCPReportBlocks() const override;
+  CallSendStatistics GetRTCPStatistics() const override;
+
+  // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
+  // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
+  // the actual processing of the audio takes place. The processing mainly
+  // consists of encoding and preparing the result for sending by adding it to a
+  // send queue.
+  // The main reason for using a task queue here is to release the native,
+  // OS-specific, audio capture thread as soon as possible to ensure that it
+  // can go back to sleep and be prepared to deliver an new captured audio
+  // packet.
+  void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) override;
+
+  int64_t GetRTT() const override;
+
+  // E2EE Custom Audio Frame Encryption
+  void SetFrameEncryptor(
+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
+
+  // Sets a frame transformer between encoder and packetizer, to transform
+  // encoded frames before sending them out the network.
+  void SetEncoderToPacketizerFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
+  // RtcpPacketTypeCounterObserver.
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override;
+
+  // ReportBlockDataObserver.
+  void OnReportBlockDataUpdated(ReportBlockData report_block) override;
+
+ private:
+  // From AudioPacketizationCallback in the ACM
+  int32_t SendData(AudioFrameType frameType,
+                   uint8_t payloadType,
+                   uint32_t rtp_timestamp,
+                   const uint8_t* payloadData,
+                   size_t payloadSize,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  bool InputMute() const;
+
+  int32_t SendRtpAudio(AudioFrameType frameType,
+                       uint8_t payloadType,
+                       uint32_t rtp_timestamp_without_offset,
+                       rtc::ArrayView<const uint8_t> payload,
+                       int64_t absolute_capture_timestamp_ms,
+                       rtc::ArrayView<const uint32_t> csrcs)
+      RTC_RUN_ON(encoder_queue_checker_);
+
+  void OnReceivedRtt(int64_t rtt_ms);
+
+  void InitFrameTransformerDelegate(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
+
+  // Thread checkers document and lock usage of some methods on voe::Channel to
+  // specific threads we know about. The goal is to eventually split up
+  // voe::Channel into parts with single-threaded semantics, and thereby reduce
+  // the need for locks.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
+  // Methods accessed from audio and video threads are checked for sequential-
+  // only access. We don't necessarily own and control these threads, so thread
+  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
+  // audio thread to another, but access is still sequential.
+  rtc::RaceChecker audio_thread_race_checker_;
+
+  mutable Mutex volume_settings_mutex_;
+
+  const uint32_t ssrc_;
+  bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false;
+
+  RtcEventLog* const event_log_;
+
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
+
+  std::unique_ptr<AudioCodingModule> audio_coding_;
+
+  // This is just an offset, RTP module will add its own random offset.
+  uint32_t timestamp_ RTC_GUARDED_BY(audio_thread_race_checker_) = 0;
+  absl::optional<int64_t> last_capture_timestamp_ms_
+      RTC_GUARDED_BY(audio_thread_race_checker_);
+
+  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_checker_);
+  bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false;
+  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_checker_) = false;
+
+  PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
+      nullptr;
+  const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
+  const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker construction_thread_;
+
+  std::atomic<bool> include_audio_level_indication_ = false;
+  std::atomic<bool> encoder_queue_is_active_ = false;
+  std::atomic<bool> first_frame_ = true;
+
+  // E2EE Audio Frame Encryption
+  rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
+      RTC_GUARDED_BY(encoder_queue_checker_);
+  // E2EE Frame Encryption Options
+  const webrtc::CryptoOptions crypto_options_;
+
+  // Delegates calls to a frame transformer to transform audio, and
+  // receives callbacks with the transformed frames; delegates calls to
+  // ChannelSend::SendRtpAudio to send the transformed audio.
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate>
+      frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_checker_);
+
+  mutable Mutex rtcp_counter_mutex_;
+  RtcpPacketTypeCounter rtcp_packet_type_counter_
+      RTC_GUARDED_BY(rtcp_counter_mutex_);
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_checker_;
+
+  SdpAudioFormat encoder_format_;
+};
+
+const int kTelephoneEventAttenuationdB = 10;
+
+class RtpPacketSenderProxy : public RtpPacketSender {
+ public:
+  RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {}
+
+  void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) {
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    MutexLock lock(&mutex_);
+    rtp_packet_pacer_ = rtp_packet_pacer;
+  }
+
+  void EnqueuePackets(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) override {
+    MutexLock lock(&mutex_);
+    rtp_packet_pacer_->EnqueuePackets(std::move(packets));
+  }
+
+  void RemovePacketsForSsrc(uint32_t ssrc) override {
+    MutexLock lock(&mutex_);
+    rtp_packet_pacer_->RemovePacketsForSsrc(ssrc);
+  }
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_;
+  Mutex mutex_;
+  RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&mutex_);
+};
+
+int32_t ChannelSend::SendData(AudioFrameType frameType,
+                              uint8_t payloadType,
+                              uint32_t rtp_timestamp,
+                              const uint8_t* payloadData,
+                              size_t payloadSize,
+                              int64_t absolute_capture_timestamp_ms) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+  rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
+  if (frame_transformer_delegate_) {
+    // Asynchronously transform the payload before sending it. After the payload
+    // is transformed, the delegate will call SendRtpAudio to send it.
+    char buf[1024];
+    rtc::SimpleStringBuilder mime_type(buf);
+    mime_type << MediaTypeToString(cricket::MEDIA_TYPE_AUDIO) << "/"
+              << encoder_format_.name;
+    frame_transformer_delegate_->Transform(
+        frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
+        payloadData, payloadSize, absolute_capture_timestamp_ms,
+        rtp_rtcp_->SSRC(), mime_type.str());
+    return 0;
+  }
+  return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
+                      absolute_capture_timestamp_ms, /*csrcs=*/{});
+}
+
+int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
+                                  uint8_t payloadType,
+                                  uint32_t rtp_timestamp_without_offset,
+                                  rtc::ArrayView<const uint8_t> payload,
+                                  int64_t absolute_capture_timestamp_ms,
+                                  rtc::ArrayView<const uint32_t> csrcs) {
+  // E2EE Custom Audio Frame Encryption (This is optional).
+  // Keep this buffer around for the lifetime of the send call.
+  rtc::Buffer encrypted_audio_payload;
+  // We don't invoke encryptor if payload is empty, which means we are to send
+  // DTMF, or the encoder entered DTX.
+  // TODO(minyue): see whether DTMF packets should be encrypted or not. In
+  // current implementation, they are not.
+  if (!payload.empty()) {
+    if (frame_encryptor_ != nullptr) {
+      // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
+      // Allocate a buffer to hold the maximum possible encrypted payload.
+      size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize(
+          cricket::MEDIA_TYPE_AUDIO, payload.size());
+      encrypted_audio_payload.SetSize(max_ciphertext_size);
+
+      // Encrypt the audio payload into the buffer.
+      size_t bytes_written = 0;
+      int encrypt_status = frame_encryptor_->Encrypt(
+          cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(),
+          /*additional_data=*/nullptr, payload, encrypted_audio_payload,
+          &bytes_written);
+      if (encrypt_status != 0) {
+        RTC_DLOG(LS_ERROR)
+            << "Channel::SendData() failed encrypt audio payload: "
+            << encrypt_status;
+        return -1;
+      }
+      // Resize the buffer to the exact number of bytes actually used.
+      encrypted_audio_payload.SetSize(bytes_written);
+      // Rewrite the payloadData and size to the new encrypted payload.
+      payload = encrypted_audio_payload;
+    } else if (crypto_options_.sframe.require_frame_encryption) {
+      RTC_DLOG(LS_ERROR) << "Channel::SendData() failed sending audio payload: "
+                            "A frame encryptor is required but one is not set.";
+      return -1;
+    }
+  }
+
+  // Push data from ACM to RTP/RTCP-module to deliver audio frame for
+  // packetization.
+  if (!rtp_rtcp_->OnSendingRtpFrame(rtp_timestamp_without_offset,
+                                    // Leaving the time when this frame was
+                                    // received from the capture device as
+                                    // undefined for voice for now.
+                                    -1, payloadType,
+                                    /*force_sender_report=*/false)) {
+    return -1;
+  }
+
+  // RTCPSender has it's own copy of the timestamp offset, added in
+  // RTCPSender::BuildSR, hence we must not add the in the offset for the above
+  // call.
+  // TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
+  // knowledge of the offset to a single place.
+
+  // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
+  RTPSenderAudio::RtpAudioFrame frame = {
+      .type = frameType,
+      .payload = payload,
+      .payload_id = payloadType,
+      .rtp_timestamp =
+          rtp_timestamp_without_offset + rtp_rtcp_->StartTimestamp(),
+      .csrcs = csrcs};
+  if (absolute_capture_timestamp_ms > 0) {
+    frame.capture_time = Timestamp::Millis(absolute_capture_timestamp_ms);
+  }
+  if (include_audio_level_indication_.load()) {
+    frame.audio_level_dbov = rms_level_.Average();
+  }
+  if (!rtp_sender_audio_->SendAudio(frame)) {
+    RTC_DLOG(LS_ERROR)
+        << "ChannelSend::SendData() failed to send data to RTP/RTCP module";
+    return -1;
+  }
+
+  return 0;
+}
+
+ChannelSend::ChannelSend(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory,
+    Transport* rtp_transport,
+    RtcpRttStats* rtcp_rtt_stats,
+    RtcEventLog* rtc_event_log,
+    FrameEncryptorInterface* frame_encryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    bool extmap_allow_mixed,
+    int rtcp_report_interval_ms,
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtpTransportControllerSendInterface* transport_controller,
+    const FieldTrialsView& field_trials)
+    : ssrc_(ssrc),
+      event_log_(rtc_event_log),
+      rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
+      retransmission_rate_limiter_(
+          new RateLimiter(clock, kMaxRetransmissionWindowMs)),
+      frame_encryptor_(frame_encryptor),
+      crypto_options_(crypto_options),
+      encoder_queue_(task_queue_factory->CreateTaskQueue(
+          "AudioEncoder",
+          TaskQueueFactory::Priority::NORMAL)),
+      encoder_queue_checker_(encoder_queue_.get()),
+      encoder_format_("x-unknown", 0, 0) {
+  audio_coding_ = AudioCodingModule::Create();
+
+  RtpRtcpInterface::Configuration configuration;
+  configuration.report_block_data_observer = this;
+  configuration.network_link_rtcp_observer =
+      transport_controller->GetRtcpObserver();
+  configuration.transport_feedback_callback =
+      transport_controller->transport_feedback_observer();
+  configuration.clock = (clock ? clock : Clock::GetRealTimeClock());
+  configuration.audio = true;
+  configuration.outgoing_transport = rtp_transport;
+
+  configuration.paced_sender = rtp_packet_pacer_proxy_.get();
+
+  configuration.event_log = event_log_;
+  configuration.rtt_stats = rtcp_rtt_stats;
+  if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) {
+    configuration.retransmission_rate_limiter =
+        retransmission_rate_limiter_.get();
+  }
+  configuration.extmap_allow_mixed = extmap_allow_mixed;
+  configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
+  configuration.rtcp_packet_type_counter_observer = this;
+
+  configuration.local_media_ssrc = ssrc;
+
+  rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
+  rtp_rtcp_->SetSendingMediaStatus(false);
+
+  rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(configuration.clock,
+                                                       rtp_rtcp_->RtpSender());
+
+  // Ensure that RTCP is enabled by default for the created channel.
+  rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
+
+  int error = audio_coding_->RegisterTransportCallback(this);
+  RTC_DCHECK_EQ(0, error);
+  if (frame_transformer)
+    InitFrameTransformerDelegate(std::move(frame_transformer));
+}
+
+ChannelSend::~ChannelSend() {
+  RTC_DCHECK(construction_thread_.IsCurrent());
+
+  // Resets the delegate's callback to ChannelSend::SendRtpAudio.
+  if (frame_transformer_delegate_)
+    frame_transformer_delegate_->Reset();
+
+  StopSend();
+  int error = audio_coding_->RegisterTransportCallback(NULL);
+  RTC_DCHECK_EQ(0, error);
+
+  // Delete the encoder task queue first to ensure that there are no running
+  // tasks when the other members are destroyed.
+  encoder_queue_ = nullptr;
+}
+
+void ChannelSend::StartSend() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK(!sending_);
+  sending_ = true;
+
+  RTC_DCHECK(packet_router_);
+  packet_router_->AddSendRtpModule(rtp_rtcp_.get(), /*remb_candidate=*/false);
+  rtp_rtcp_->SetSendingMediaStatus(true);
+  int ret = rtp_rtcp_->SetSendingStatus(true);
+  RTC_DCHECK_EQ(0, ret);
+
+  // It is now OK to start processing on the encoder task queue.
+  first_frame_.store(true);
+  encoder_queue_is_active_.store(true);
+}
+
+void ChannelSend::StopSend() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!sending_) {
+    return;
+  }
+  sending_ = false;
+  encoder_queue_is_active_.store(false);
+
+  // Wait until all pending encode tasks are executed and clear any remaining
+  // buffers in the encoder.
+  rtc::Event flush;
+  encoder_queue_->PostTask([this, &flush]() {
+    RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+    CallEncoder([](AudioEncoder* encoder) { encoder->Reset(); });
+    flush.Set();
+  });
+  flush.Wait(rtc::Event::kForever);
+
+  // Reset sending SSRC and sequence number and triggers direct transmission
+  // of RTCP BYE
+  if (rtp_rtcp_->SetSendingStatus(false) == -1) {
+    RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
+  }
+  rtp_rtcp_->SetSendingMediaStatus(false);
+
+  RTC_DCHECK(packet_router_);
+  packet_router_->RemoveSendRtpModule(rtp_rtcp_.get());
+  rtp_packet_pacer_proxy_->RemovePacketsForSsrc(rtp_rtcp_->SSRC());
+}
+
+void ChannelSend::SetEncoder(int payload_type,
+                             const SdpAudioFormat& encoder_format,
+                             std::unique_ptr<AudioEncoder> encoder) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_GE(payload_type, 0);
+  RTC_DCHECK_LE(payload_type, 127);
+
+  // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
+  // as well as some other things, so we collect this info and send it along.
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type,
+                                          encoder->RtpTimestampRateHz());
+  rtp_sender_audio_->RegisterAudioPayload("audio", payload_type,
+                                          encoder->RtpTimestampRateHz(),
+                                          encoder->NumChannels(), 0);
+
+  encoder_format_ = encoder_format;
+  audio_coding_->SetEncoder(std::move(encoder));
+}
+
+void ChannelSend::ModifyEncoder(
+    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
+  // This method can be called on the worker thread, module process thread
+  // or network thread. Audio coding is thread safe, so we do not need to
+  // enforce the calling thread.
+  audio_coding_->ModifyEncoder(modifier);
+}
+
+void ChannelSend::CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) {
+  ModifyEncoder([modifier](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+    if (*encoder_ptr) {
+      modifier(encoder_ptr->get());
+    } else {
+      RTC_DLOG(LS_WARNING) << "Trying to call unset encoder.";
+    }
+  });
+}
+
+void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
+  // This method can be called on the worker thread, module process thread
+  // or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
+  // TODO(solenberg): Figure out a good way to check this or enforce calling
+  // rules.
+  // RTC_DCHECK(worker_thread_checker_.IsCurrent() ||
+  //            module_process_thread_checker_.IsCurrent());
+  CallEncoder([&](AudioEncoder* encoder) {
+    encoder->OnReceivedUplinkAllocation(update);
+  });
+  retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
+}
+
+int ChannelSend::GetTargetBitrate() const {
+  return audio_coding_->GetTargetBitrate();
+}
+
+void ChannelSend::OnReportBlockDataUpdated(ReportBlockData report_block) {
+  float packet_loss_rate = report_block.fraction_lost();
+  CallEncoder([&](AudioEncoder* encoder) {
+    encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
+  });
+}
+
+void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+
+  // Deliver RTCP packet to RTP/RTCP module for parsing
+  rtp_rtcp_->IncomingRtcpPacket(rtc::MakeArrayView(data, length));
+
+  int64_t rtt = GetRTT();
+  if (rtt == 0) {
+    // Waiting for valid RTT.
+    return;
+  }
+
+  int64_t nack_window_ms = rtt;
+  if (nack_window_ms < kMinRetransmissionWindowMs) {
+    nack_window_ms = kMinRetransmissionWindowMs;
+  } else if (nack_window_ms > kMaxRetransmissionWindowMs) {
+    nack_window_ms = kMaxRetransmissionWindowMs;
+  }
+  retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
+
+  OnReceivedRtt(rtt);
+}
+
+void ChannelSend::SetInputMute(bool enable) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  MutexLock lock(&volume_settings_mutex_);
+  input_mute_ = enable;
+}
+
+bool ChannelSend::InputMute() const {
+  MutexLock lock(&volume_settings_mutex_);
+  return input_mute_;
+}
+
+bool ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_LE(0, event);
+  RTC_DCHECK_GE(255, event);
+  RTC_DCHECK_LE(0, duration_ms);
+  RTC_DCHECK_GE(65535, duration_ms);
+  if (!sending_) {
+    return false;
+  }
+  if (rtp_sender_audio_->SendTelephoneEvent(
+          event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
+    RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
+    return false;
+  }
+  return true;
+}
+
+void ChannelSend::RegisterCngPayloadType(int payload_type,
+                                         int payload_frequency) {
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
+  rtp_sender_audio_->RegisterAudioPayload("CN", payload_type, payload_frequency,
+                                          1, 0);
+}
+
+void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
+                                                   int payload_frequency) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_LE(0, payload_type);
+  RTC_DCHECK_GE(127, payload_type);
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
+  rtp_sender_audio_->RegisterAudioPayload("telephone-event", payload_type,
+                                          payload_frequency, 0, 0);
+}
+
+void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  include_audio_level_indication_.store(enable);
+  if (enable) {
+    rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevel::Uri(), id);
+  } else {
+    rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevel::Uri());
+  }
+}
+
+void ChannelSend::RegisterSenderCongestionControlObjects(
+    RtpTransportControllerSendInterface* transport) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RtpPacketSender* rtp_packet_pacer = transport->packet_sender();
+  PacketRouter* packet_router = transport->packet_router();
+
+  RTC_DCHECK(rtp_packet_pacer);
+  RTC_DCHECK(packet_router);
+  RTC_DCHECK(!packet_router_);
+  rtp_packet_pacer_proxy_->SetPacketPacer(rtp_packet_pacer);
+  rtp_rtcp_->SetStorePacketsStatus(true, 600);
+  packet_router_ = packet_router;
+}
+
+void ChannelSend::ResetSenderCongestionControlObjects() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK(packet_router_);
+  rtp_rtcp_->SetStorePacketsStatus(false, 600);
+  packet_router_ = nullptr;
+  rtp_packet_pacer_proxy_->SetPacketPacer(nullptr);
+}
+
+void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Note: SetCNAME() accepts a c string of length at most 255.
+  const std::string c_name_limited(c_name.substr(0, 255));
+  int ret = rtp_rtcp_->SetCNAME(c_name_limited.c_str()) != 0;
+  RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME";
+}
+
+std::vector<ReportBlockData> ChannelSend::GetRemoteRTCPReportBlocks() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Get the report blocks from the latest received RTCP Sender or Receiver
+  // Report. Each element in the vector contains the sender's SSRC and a
+  // report block according to RFC 3550.
+  return rtp_rtcp_->GetLatestReportBlockData();
+}
+
+CallSendStatistics ChannelSend::GetRTCPStatistics() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  CallSendStatistics stats = {0};
+  stats.rttMs = GetRTT();
+
+  StreamDataCounters rtp_stats;
+  StreamDataCounters rtx_stats;
+  rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
+  stats.payload_bytes_sent =
+      rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
+  stats.header_and_padding_bytes_sent =
+      rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
+      rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
+
+  // TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
+  // separate outbound-rtp stream objects.
+  stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
+  stats.packetsSent =
+      rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+  stats.total_packet_send_delay = rtp_stats.transmitted.total_packet_delay;
+  stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
+  stats.report_block_datas = rtp_rtcp_->GetLatestReportBlockData();
+
+  {
+    MutexLock lock(&rtcp_counter_mutex_);
+    stats.nacks_received = rtcp_packet_type_counter_.nack_packets;
+  }
+
+  return stats;
+}
+
+void ChannelSend::RtcpPacketTypesCounterUpdated(
+    uint32_t ssrc,
+    const RtcpPacketTypeCounter& packet_counter) {
+  if (ssrc != ssrc_) {
+    return;
+  }
+  MutexLock lock(&rtcp_counter_mutex_);
+  rtcp_packet_type_counter_ = packet_counter;
+}
+
+void ChannelSend::ProcessAndEncodeAudio(
+    std::unique_ptr<AudioFrame> audio_frame) {
+  TRACE_EVENT0("webrtc", "ChannelSend::ProcessAndEncodeAudio");
+
+  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  RTC_DCHECK_LE(audio_frame->num_channels_, 8);
+
+  if (!encoder_queue_is_active_.load()) {
+    return;
+  }
+
+  // Update `timestamp_` based on the capture timestamp for the first frame
+  // after sending is resumed.
+  if (first_frame_.load()) {
+    first_frame_.store(false);
+    if (last_capture_timestamp_ms_ &&
+        audio_frame->absolute_capture_timestamp_ms()) {
+      int64_t diff_ms = *audio_frame->absolute_capture_timestamp_ms() -
+                        *last_capture_timestamp_ms_;
+      // Truncate to whole frames and subtract one since `timestamp_` was
+      // incremented after the last frame.
+      int64_t diff_frames = diff_ms * audio_frame->sample_rate_hz() / 1000 /
+                                audio_frame->samples_per_channel() -
+                            1;
+      timestamp_ += std::max<int64_t>(
+          diff_frames * audio_frame->samples_per_channel(), 0);
+    }
+  }
+
+  audio_frame->timestamp_ = timestamp_;
+  timestamp_ += audio_frame->samples_per_channel_;
+  last_capture_timestamp_ms_ = audio_frame->absolute_capture_timestamp_ms();
+
+  // Profile time between when the audio frame is added to the task queue and
+  // when the task is actually executed.
+  audio_frame->UpdateProfileTimeStamp();
+  encoder_queue_->PostTask(
+      [this, audio_frame = std::move(audio_frame)]() mutable {
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+        if (!encoder_queue_is_active_.load()) {
+          return;
+        }
+        // Measure time between when the audio frame is added to the task queue
+        // and when the task is actually executed. Goal is to keep track of
+        // unwanted extra latency added by the task queue.
+        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs",
+                                   audio_frame->ElapsedProfileTimeMs());
+
+        bool is_muted = InputMute();
+        AudioFrameOperations::Mute(audio_frame.get(), previous_frame_muted_,
+                                   is_muted);
+
+        if (include_audio_level_indication_.load()) {
+          size_t length =
+              audio_frame->samples_per_channel_ * audio_frame->num_channels_;
+          RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes);
+          if (is_muted && previous_frame_muted_) {
+            rms_level_.AnalyzeMuted(length);
+          } else {
+            rms_level_.Analyze(
+                rtc::ArrayView<const int16_t>(audio_frame->data(), length));
+          }
+        }
+        previous_frame_muted_ = is_muted;
+
+        // This call will trigger AudioPacketizationCallback::SendData if
+        // encoding is done and payload is ready for packetization and
+        // transmission. Otherwise, it will return without invoking the
+        // callback.
+        if (audio_coding_->Add10MsData(*audio_frame) < 0) {
+          RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
+          return;
+        }
+      });
+}
+
+ANAStats ChannelSend::GetANAStatistics() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return audio_coding_->GetANAStats();
+}
+
+RtpRtcpInterface* ChannelSend::GetRtpRtcp() const {
+  return rtp_rtcp_.get();
+}
+
+int64_t ChannelSend::GetRTT() const {
+  std::vector<ReportBlockData> report_blocks =
+      rtp_rtcp_->GetLatestReportBlockData();
+  if (report_blocks.empty()) {
+    return 0;
+  }
+
+  // We don't know in advance the remote ssrc used by the other end's receiver
+  // reports, so use the first report block for the RTT.
+  return report_blocks.front().last_rtt().ms();
+}
+
+void ChannelSend::SetFrameEncryptor(
+    rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  encoder_queue_->PostTask([this, frame_encryptor]() mutable {
+    RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+    frame_encryptor_ = std::move(frame_encryptor);
+  });
+}
+
+void ChannelSend::SetEncoderToPacketizerFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!frame_transformer)
+    return;
+
+  encoder_queue_->PostTask(
+      [this, frame_transformer = std::move(frame_transformer)]() mutable {
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+        InitFrameTransformerDelegate(std::move(frame_transformer));
+      });
+}
+
+void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
+  // Invoke audio encoders OnReceivedRtt().
+  CallEncoder(
+      [rtt_ms](AudioEncoder* encoder) { encoder->OnReceivedRtt(rtt_ms); });
+}
+
+void ChannelSend::InitFrameTransformerDelegate(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+  RTC_DCHECK(frame_transformer);
+  RTC_DCHECK(!frame_transformer_delegate_);
+
+  // Pass a callback to ChannelSend::SendRtpAudio, to be called by the delegate
+  // to send the transformed audio.
+  ChannelSendFrameTransformerDelegate::SendFrameCallback send_audio_callback =
+      [this](AudioFrameType frameType, uint8_t payloadType,
+             uint32_t rtp_timestamp_with_offset,
+             rtc::ArrayView<const uint8_t> payload,
+             int64_t absolute_capture_timestamp_ms,
+             rtc::ArrayView<const uint32_t> csrcs) {
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+        return SendRtpAudio(
+            frameType, payloadType,
+            rtp_timestamp_with_offset - rtp_rtcp_->StartTimestamp(), payload,
+            absolute_capture_timestamp_ms, csrcs);
+      };
+  frame_transformer_delegate_ =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          std::move(send_audio_callback), std::move(frame_transformer),
+          encoder_queue_.get());
+  frame_transformer_delegate_->Init();
+}
+
+}  // namespace
+
+std::unique_ptr<ChannelSendInterface> CreateChannelSend(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory,
+    Transport* rtp_transport,
+    RtcpRttStats* rtcp_rtt_stats,
+    RtcEventLog* rtc_event_log,
+    FrameEncryptorInterface* frame_encryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    bool extmap_allow_mixed,
+    int rtcp_report_interval_ms,
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtpTransportControllerSendInterface* transport_controller,
+    const FieldTrialsView& field_trials) {
+  return std::make_unique<ChannelSend>(
+      clock, task_queue_factory, rtp_transport, rtcp_rtt_stats, rtc_event_log,
+      frame_encryptor, crypto_options, extmap_allow_mixed,
+      rtcp_report_interval_ms, ssrc, std::move(frame_transformer),
+      transport_controller, field_trials);
+}
+
+}  // namespace voe
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/channel_send.h

@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CHANNEL_SEND_H_
+#define AUDIO_CHANNEL_SEND_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/crypto/crypto_options.h"
+#include "api/field_trials_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/function_view.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+
+namespace webrtc {
+
+class FrameEncryptorInterface;
+class RtcEventLog;
+class RtpTransportControllerSendInterface;
+
+struct CallSendStatistics {
+  int64_t rttMs;
+  int64_t payload_bytes_sent;
+  int64_t header_and_padding_bytes_sent;
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
+  uint64_t retransmitted_bytes_sent;
+  int packetsSent;
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
+  TimeDelta total_packet_send_delay = TimeDelta::Zero();
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
+  uint64_t retransmitted_packets_sent;
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the sender-source SSRC pair is unique and per-pair the
+  // ReportBlockData represents the latest Report Block that was received for
+  // that pair.
+  std::vector<ReportBlockData> report_block_datas;
+  uint32_t nacks_received;
+};
+
+namespace voe {
+
+class ChannelSendInterface {
+ public:
+  virtual ~ChannelSendInterface() = default;
+
+  virtual void ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
+
+  virtual CallSendStatistics GetRTCPStatistics() const = 0;
+
+  virtual void SetEncoder(int payload_type,
+                          const SdpAudioFormat& encoder_format,
+                          std::unique_ptr<AudioEncoder> encoder) = 0;
+  virtual void ModifyEncoder(
+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
+  virtual void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) = 0;
+
+  // Use 0 to indicate that the extension should not be registered.
+  virtual void SetRTCP_CNAME(absl::string_view c_name) = 0;
+  virtual void SetSendAudioLevelIndicationStatus(bool enable, int id) = 0;
+  virtual void RegisterSenderCongestionControlObjects(
+      RtpTransportControllerSendInterface* transport) = 0;
+  virtual void ResetSenderCongestionControlObjects() = 0;
+  virtual std::vector<ReportBlockData> GetRemoteRTCPReportBlocks() const = 0;
+  virtual ANAStats GetANAStatistics() const = 0;
+  virtual void RegisterCngPayloadType(int payload_type,
+                                      int payload_frequency) = 0;
+  virtual void SetSendTelephoneEventPayloadType(int payload_type,
+                                                int payload_frequency) = 0;
+  virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
+  virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
+  virtual int GetTargetBitrate() const = 0;
+  virtual void SetInputMute(bool muted) = 0;
+
+  virtual void ProcessAndEncodeAudio(
+      std::unique_ptr<AudioFrame> audio_frame) = 0;
+  virtual RtpRtcpInterface* GetRtpRtcp() const = 0;
+
+  // In RTP we currently rely on RTCP packets (`ReceivedRTCPPacket`) to inform
+  // about RTT.
+  // In media transport we rely on the TargetTransferRateObserver instead.
+  // In other words, if you are using RTP, you should expect
+  // `ReceivedRTCPPacket` to be called, if you are using media transport,
+  // `OnTargetTransferRate` will be called.
+  //
+  // In future, RTP media will move to the media transport implementation and
+  // these conditions will be removed.
+  // Returns the RTT in milliseconds.
+  virtual int64_t GetRTT() const = 0;
+  virtual void StartSend() = 0;
+  virtual void StopSend() = 0;
+
+  // E2EE Custom Audio Frame Encryption (Optional)
+  virtual void SetFrameEncryptor(
+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
+
+  // Sets a frame transformer between encoder and packetizer, to transform
+  // encoded frames before sending them out the network.
+  virtual void SetEncoderToPacketizerFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+          frame_transformer) = 0;
+};
+
+std::unique_ptr<ChannelSendInterface> CreateChannelSend(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory,
+    Transport* rtp_transport,
+    RtcpRttStats* rtcp_rtt_stats,
+    RtcEventLog* rtc_event_log,
+    FrameEncryptorInterface* frame_encryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    bool extmap_allow_mixed,
+    int rtcp_report_interval_ms,
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtpTransportControllerSendInterface* transport_controller,
+    const FieldTrialsView& field_trials);
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_CHANNEL_SEND_H_

TMessagesProj/jni/voip/webrtc/audio/channel_send_frame_transformer_delegate.cc

@@ -0,0 +1,213 @@
+/*
+ *  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 "audio/channel_send_frame_transformer_delegate.h"
+
+#include <utility>
+#include <vector>
+
+namespace webrtc {
+namespace {
+
+using IfaceFrameType = TransformableAudioFrameInterface::FrameType;
+
+IfaceFrameType InternalFrameTypeToInterfaceFrameType(
+    const AudioFrameType frame_type) {
+  switch (frame_type) {
+    case AudioFrameType::kEmptyFrame:
+      return IfaceFrameType::kEmptyFrame;
+    case AudioFrameType::kAudioFrameSpeech:
+      return IfaceFrameType::kAudioFrameSpeech;
+    case AudioFrameType::kAudioFrameCN:
+      return IfaceFrameType::kAudioFrameCN;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return IfaceFrameType::kEmptyFrame;
+}
+
+AudioFrameType InterfaceFrameTypeToInternalFrameType(
+    const IfaceFrameType frame_type) {
+  switch (frame_type) {
+    case IfaceFrameType::kEmptyFrame:
+      return AudioFrameType::kEmptyFrame;
+    case IfaceFrameType::kAudioFrameSpeech:
+      return AudioFrameType::kAudioFrameSpeech;
+    case IfaceFrameType::kAudioFrameCN:
+      return AudioFrameType::kAudioFrameCN;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return AudioFrameType::kEmptyFrame;
+}
+
+class TransformableOutgoingAudioFrame
+    : public TransformableAudioFrameInterface {
+ public:
+  TransformableOutgoingAudioFrame(
+      AudioFrameType frame_type,
+      uint8_t payload_type,
+      uint32_t rtp_timestamp_with_offset,
+      const uint8_t* payload_data,
+      size_t payload_size,
+      absl::optional<uint64_t> absolute_capture_timestamp_ms,
+      uint32_t ssrc,
+      std::vector<uint32_t> csrcs,
+      const std::string& codec_mime_type,
+      absl::optional<uint16_t> sequence_number)
+      : frame_type_(frame_type),
+        payload_type_(payload_type),
+        rtp_timestamp_with_offset_(rtp_timestamp_with_offset),
+        payload_(payload_data, payload_size),
+        absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
+        ssrc_(ssrc),
+        csrcs_(std::move(csrcs)),
+        codec_mime_type_(codec_mime_type),
+        sequence_number_(sequence_number) {}
+  ~TransformableOutgoingAudioFrame() override = default;
+  rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
+  void SetData(rtc::ArrayView<const uint8_t> data) override {
+    payload_.SetData(data.data(), data.size());
+  }
+  uint32_t GetTimestamp() const override { return rtp_timestamp_with_offset_; }
+  uint32_t GetSsrc() const override { return ssrc_; }
+
+  IfaceFrameType Type() const override {
+    return InternalFrameTypeToInterfaceFrameType(frame_type_);
+  }
+
+  uint8_t GetPayloadType() const override { return payload_type_; }
+  Direction GetDirection() const override { return Direction::kSender; }
+  std::string GetMimeType() const override { return codec_mime_type_; }
+
+  rtc::ArrayView<const uint32_t> GetContributingSources() const override {
+    return csrcs_;
+  }
+
+  const absl::optional<uint16_t> SequenceNumber() const override {
+    return sequence_number_;
+  }
+
+  void SetRTPTimestamp(uint32_t rtp_timestamp_with_offset) override {
+    rtp_timestamp_with_offset_ = rtp_timestamp_with_offset;
+  }
+
+  absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
+    return absolute_capture_timestamp_ms_;
+  }
+
+ private:
+  AudioFrameType frame_type_;
+  uint8_t payload_type_;
+  uint32_t rtp_timestamp_with_offset_;
+  rtc::Buffer payload_;
+  absl::optional<uint64_t> absolute_capture_timestamp_ms_;
+  uint32_t ssrc_;
+  std::vector<uint32_t> csrcs_;
+  std::string codec_mime_type_;
+  absl::optional<uint16_t> sequence_number_;
+};
+}  // namespace
+
+ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
+    SendFrameCallback send_frame_callback,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    TaskQueueBase* encoder_queue)
+    : send_frame_callback_(send_frame_callback),
+      frame_transformer_(std::move(frame_transformer)),
+      encoder_queue_(encoder_queue) {}
+
+void ChannelSendFrameTransformerDelegate::Init() {
+  frame_transformer_->RegisterTransformedFrameCallback(
+      rtc::scoped_refptr<TransformedFrameCallback>(this));
+}
+
+void ChannelSendFrameTransformerDelegate::Reset() {
+  frame_transformer_->UnregisterTransformedFrameCallback();
+  frame_transformer_ = nullptr;
+
+  MutexLock lock(&send_lock_);
+  send_frame_callback_ = SendFrameCallback();
+}
+
+void ChannelSendFrameTransformerDelegate::Transform(
+    AudioFrameType frame_type,
+    uint8_t payload_type,
+    uint32_t rtp_timestamp,
+    const uint8_t* payload_data,
+    size_t payload_size,
+    int64_t absolute_capture_timestamp_ms,
+    uint32_t ssrc,
+    const std::string& codec_mimetype) {
+  {
+    MutexLock lock(&send_lock_);
+    if (short_circuit_) {
+      send_frame_callback_(
+          frame_type, payload_type, rtp_timestamp,
+          rtc::ArrayView<const uint8_t>(payload_data, payload_size),
+          absolute_capture_timestamp_ms, /*csrcs=*/{});
+      return;
+    }
+  }
+  frame_transformer_->Transform(
+      std::make_unique<TransformableOutgoingAudioFrame>(
+          frame_type, payload_type, rtp_timestamp, payload_data, payload_size,
+          absolute_capture_timestamp_ms, ssrc,
+          /*csrcs=*/std::vector<uint32_t>(), codec_mimetype,
+          /*sequence_number=*/absl::nullopt));
+}
+
+void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) {
+  MutexLock lock(&send_lock_);
+  if (!send_frame_callback_)
+    return;
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate(this);
+  encoder_queue_->PostTask(
+      [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
+        delegate->SendFrame(std::move(frame));
+      });
+}
+
+void ChannelSendFrameTransformerDelegate::StartShortCircuiting() {
+  MutexLock lock(&send_lock_);
+  short_circuit_ = true;
+}
+
+void ChannelSendFrameTransformerDelegate::SendFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) const {
+  MutexLock lock(&send_lock_);
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  if (!send_frame_callback_)
+    return;
+  auto* transformed_frame =
+      static_cast<TransformableAudioFrameInterface*>(frame.get());
+  send_frame_callback_(
+      InterfaceFrameTypeToInternalFrameType(transformed_frame->Type()),
+      transformed_frame->GetPayloadType(), transformed_frame->GetTimestamp(),
+      transformed_frame->GetData(),
+      transformed_frame->AbsoluteCaptureTimestamp()
+          ? *transformed_frame->AbsoluteCaptureTimestamp()
+          : 0,
+      transformed_frame->GetContributingSources());
+}
+
+std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
+    TransformableAudioFrameInterface* original) {
+  std::vector<uint32_t> csrcs;
+  csrcs.assign(original->GetContributingSources().begin(),
+               original->GetContributingSources().end());
+  return std::make_unique<TransformableOutgoingAudioFrame>(
+      InterfaceFrameTypeToInternalFrameType(original->Type()),
+      original->GetPayloadType(), original->GetTimestamp(),
+      original->GetData().data(), original->GetData().size(),
+      original->AbsoluteCaptureTimestamp(), original->GetSsrc(),
+      std::move(csrcs), original->GetMimeType(), original->SequenceNumber());
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/channel_send_frame_transformer_delegate.h

@@ -0,0 +1,90 @@
+/*
+ *  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 AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
+#define AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
+
+#include <memory>
+#include <string>
+
+#include "api/frame_transformer_interface.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// Delegates calls to FrameTransformerInterface to transform frames, and to
+// ChannelSend to send the transformed frames using `send_frame_callback_` on
+// the `encoder_queue_`.
+// OnTransformedFrame() can be called from any thread, the delegate ensures
+// thread-safe access to the ChannelSend callback.
+class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
+ public:
+  using SendFrameCallback =
+      std::function<int32_t(AudioFrameType frameType,
+                            uint8_t payloadType,
+                            uint32_t rtp_timestamp_with_offset,
+                            rtc::ArrayView<const uint8_t> payload,
+                            int64_t absolute_capture_timestamp_ms,
+                            rtc::ArrayView<const uint32_t> csrcs)>;
+  ChannelSendFrameTransformerDelegate(
+      SendFrameCallback send_frame_callback,
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+      TaskQueueBase* encoder_queue);
+
+  // Registers `this` as callback for `frame_transformer_`, to get the
+  // transformed frames.
+  void Init();
+
+  // Unregisters and releases the `frame_transformer_` reference, and resets
+  // `send_frame_callback_` under lock. Called from ChannelSend destructor to
+  // prevent running the callback on a dangling channel.
+  void Reset();
+
+  // Delegates the call to FrameTransformerInterface::TransformFrame, to
+  // transform the frame asynchronously.
+  void Transform(AudioFrameType frame_type,
+                 uint8_t payload_type,
+                 uint32_t rtp_timestamp,
+                 const uint8_t* payload_data,
+                 size_t payload_size,
+                 int64_t absolute_capture_timestamp_ms,
+                 uint32_t ssrc,
+                 const std::string& codec_mime_type);
+
+  // Implements TransformedFrameCallback. Can be called on any thread.
+  void OnTransformedFrame(
+      std::unique_ptr<TransformableFrameInterface> frame) override;
+
+  void StartShortCircuiting() override;
+
+  // Delegates the call to ChannelSend::SendRtpAudio on the `encoder_queue_`,
+  // by calling `send_audio_callback_`.
+  void SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
+
+ protected:
+  ~ChannelSendFrameTransformerDelegate() override = default;
+
+ private:
+  mutable Mutex send_lock_;
+  SendFrameCallback send_frame_callback_ RTC_GUARDED_BY(send_lock_);
+  rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
+  TaskQueueBase* const encoder_queue_;
+  bool short_circuit_ RTC_GUARDED_BY(send_lock_) = false;
+};
+
+std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
+    TransformableAudioFrameInterface* original);
+
+}  // namespace webrtc
+#endif  // AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_

TMessagesProj/jni/voip/webrtc/audio/conversion.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CONVERSION_H_
+#define AUDIO_CONVERSION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// Convert fixed point number with 8 bit fractional part, to floating point.
+inline float Q8ToFloat(uint32_t v) {
+  return static_cast<float>(v) / (1 << 8);
+}
+
+// Convert fixed point number with 14 bit fractional part, to floating point.
+inline float Q14ToFloat(uint32_t v) {
+  return static_cast<float>(v) / (1 << 14);
+}
+}  // namespace webrtc
+
+#endif  // AUDIO_CONVERSION_H_

TMessagesProj/jni/voip/webrtc/audio/mock_voe_channel_proxy.h

@@ -0,0 +1,188 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
+#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/test/mock_frame_encryptor.h"
+#include "audio/channel_receive.h"
+#include "audio/channel_send.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockChannelReceive : public voe::ChannelReceiveInterface {
+ public:
+  MOCK_METHOD(void, SetNACKStatus, (bool enable, int max_packets), (override));
+  MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
+  MOCK_METHOD(void,
+              RegisterReceiverCongestionControlObjects,
+              (PacketRouter*),
+              (override));
+  MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override));
+  MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override));
+  MOCK_METHOD(NetworkStatistics,
+              GetNetworkStatistics,
+              (bool),
+              (const, override));
+  MOCK_METHOD(AudioDecodingCallStats,
+              GetDecodingCallStatistics,
+              (),
+              (const, override));
+  MOCK_METHOD(int, GetSpeechOutputLevelFullRange, (), (const, override));
+  MOCK_METHOD(double, GetTotalOutputEnergy, (), (const, override));
+  MOCK_METHOD(double, GetTotalOutputDuration, (), (const, override));
+  MOCK_METHOD(uint32_t, GetDelayEstimate, (), (const, override));
+  MOCK_METHOD(void, SetSink, (AudioSinkInterface*), (override));
+  MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived& packet), (override));
+  MOCK_METHOD(void,
+              ReceivedRTCPPacket,
+              (const uint8_t*, size_t length),
+              (override));
+  MOCK_METHOD(void, SetChannelOutputVolumeScaling, (float scaling), (override));
+  MOCK_METHOD(AudioMixer::Source::AudioFrameInfo,
+              GetAudioFrameWithInfo,
+              (int sample_rate_hz, AudioFrame*),
+              (override));
+  MOCK_METHOD(int, PreferredSampleRate, (), (const, override));
+  MOCK_METHOD(void, SetSourceTracker, (SourceTracker*), (override));
+  MOCK_METHOD(void,
+              SetAssociatedSendChannel,
+              (const voe::ChannelSendInterface*),
+              (override));
+  MOCK_METHOD(bool,
+              GetPlayoutRtpTimestamp,
+              (uint32_t*, int64_t*),
+              (const, override));
+  MOCK_METHOD(void,
+              SetEstimatedPlayoutNtpTimestampMs,
+              (int64_t ntp_timestamp_ms, int64_t time_ms),
+              (override));
+  MOCK_METHOD(absl::optional<int64_t>,
+              GetCurrentEstimatedPlayoutNtpTimestampMs,
+              (int64_t now_ms),
+              (const, override));
+  MOCK_METHOD(absl::optional<Syncable::Info>,
+              GetSyncInfo,
+              (),
+              (const, override));
+  MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override));
+  MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override));
+  MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>),
+              GetReceiveCodec,
+              (),
+              (const, override));
+  MOCK_METHOD(void,
+              SetReceiveCodecs,
+              ((const std::map<int, SdpAudioFormat>& codecs)),
+              (override));
+  MOCK_METHOD(void, StartPlayout, (), (override));
+  MOCK_METHOD(void, StopPlayout, (), (override));
+  MOCK_METHOD(
+      void,
+      SetDepacketizerToDecoderFrameTransformer,
+      (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
+      (override));
+  MOCK_METHOD(
+      void,
+      SetFrameDecryptor,
+      (rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
+      (override));
+  MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override));
+  MOCK_METHOD(uint32_t, GetLocalSsrc, (), (const, override));
+};
+
+class MockChannelSend : public voe::ChannelSendInterface {
+ public:
+  MOCK_METHOD(void,
+              SetEncoder,
+              (int payload_type,
+               const SdpAudioFormat& encoder_format,
+               std::unique_ptr<AudioEncoder> encoder),
+              (override));
+  MOCK_METHOD(
+      void,
+      ModifyEncoder,
+      (rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier),
+      (override));
+  MOCK_METHOD(void,
+              CallEncoder,
+              (rtc::FunctionView<void(AudioEncoder*)> modifier),
+              (override));
+  MOCK_METHOD(void, SetRTCP_CNAME, (absl::string_view c_name), (override));
+  MOCK_METHOD(void,
+              SetSendAudioLevelIndicationStatus,
+              (bool enable, int id),
+              (override));
+  MOCK_METHOD(void,
+              RegisterSenderCongestionControlObjects,
+              (RtpTransportControllerSendInterface*),
+              (override));
+  MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override));
+  MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override));
+  MOCK_METHOD(std::vector<ReportBlockData>,
+              GetRemoteRTCPReportBlocks,
+              (),
+              (const, override));
+  MOCK_METHOD(ANAStats, GetANAStatistics, (), (const, override));
+  MOCK_METHOD(void,
+              RegisterCngPayloadType,
+              (int payload_type, int payload_frequency),
+              (override));
+  MOCK_METHOD(void,
+              SetSendTelephoneEventPayloadType,
+              (int payload_type, int payload_frequency),
+              (override));
+  MOCK_METHOD(bool,
+              SendTelephoneEventOutband,
+              (int event, int duration_ms),
+              (override));
+  MOCK_METHOD(void,
+              OnBitrateAllocation,
+              (BitrateAllocationUpdate update),
+              (override));
+  MOCK_METHOD(void, SetInputMute, (bool muted), (override));
+  MOCK_METHOD(void,
+              ReceivedRTCPPacket,
+              (const uint8_t*, size_t length),
+              (override));
+  MOCK_METHOD(void,
+              ProcessAndEncodeAudio,
+              (std::unique_ptr<AudioFrame>),
+              (override));
+  MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override));
+  MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
+  MOCK_METHOD(int64_t, GetRTT, (), (const, override));
+  MOCK_METHOD(void, StartSend, (), (override));
+  MOCK_METHOD(void, StopSend, (), (override));
+  MOCK_METHOD(void,
+              SetFrameEncryptor,
+              (rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor),
+              (override));
+  MOCK_METHOD(
+      void,
+      SetEncoderToPacketizerFrameTransformer,
+      (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
+      (override));
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_

TMessagesProj/jni/voip/webrtc/audio/remix_resample.cc

@@ -0,0 +1,91 @@
+/*
+ *  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 "audio/remix_resample.h"
+
+#include "api/audio/audio_frame.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace voe {
+
+void RemixAndResample(const AudioFrame& src_frame,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame) {
+  RemixAndResample(src_frame.data(), src_frame.samples_per_channel_,
+                   src_frame.num_channels_, src_frame.sample_rate_hz_,
+                   resampler, dst_frame);
+  dst_frame->timestamp_ = src_frame.timestamp_;
+  dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
+  dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
+  dst_frame->packet_infos_ = src_frame.packet_infos_;
+}
+
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      size_t num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame) {
+  const int16_t* audio_ptr = src_data;
+  size_t audio_ptr_num_channels = num_channels;
+  int16_t downmixed_audio[AudioFrame::kMaxDataSizeSamples];
+
+  // Downmix before resampling.
+  if (num_channels > dst_frame->num_channels_) {
+    RTC_DCHECK(num_channels == 2 || num_channels == 4)
+        << "num_channels: " << num_channels;
+    RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
+        << "dst_frame->num_channels_: " << dst_frame->num_channels_;
+
+    AudioFrameOperations::DownmixChannels(
+        src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
+        downmixed_audio);
+    audio_ptr = downmixed_audio;
+    audio_ptr_num_channels = dst_frame->num_channels_;
+  }
+
+  if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
+                                    audio_ptr_num_channels) == -1) {
+    RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
+                << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
+                << dst_frame->sample_rate_hz_
+                << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
+  }
+
+  // TODO(yujo): for muted input frames, don't resample. Either 1) allow
+  // resampler to return output length without doing the resample, so we know
+  // how much to zero here; or 2) make resampler accept a hint that the input is
+  // zeroed.
+  const size_t src_length = samples_per_channel * audio_ptr_num_channels;
+  int out_length =
+      resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
+                          AudioFrame::kMaxDataSizeSamples);
+  if (out_length == -1) {
+    RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
+                << ", src_length = " << src_length
+                << ", dst_frame->mutable_data() = "
+                << dst_frame->mutable_data();
+  }
+  dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
+
+  // Upmix after resampling.
+  if (num_channels == 1 && dst_frame->num_channels_ == 2) {
+    // The audio in dst_frame really is mono at this point; MonoToStereo will
+    // set this back to stereo.
+    dst_frame->num_channels_ = 1;
+    AudioFrameOperations::UpmixChannels(2, dst_frame);
+  }
+}
+
+}  // namespace voe
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/remix_resample.h

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_REMIX_RESAMPLE_H_
+#define AUDIO_REMIX_RESAMPLE_H_
+
+#include "api/audio/audio_frame.h"
+#include "common_audio/resampler/include/push_resampler.h"
+
+namespace webrtc {
+namespace voe {
+
+// Upmix or downmix and resample the audio to `dst_frame`. Expects `dst_frame`
+// to have its sample rate and channels members set to the desired values.
+// Updates the `samples_per_channel_` member accordingly.
+//
+// This version has an AudioFrame `src_frame` as input and sets the output
+// `timestamp_`, `elapsed_time_ms_` and `ntp_time_ms_` members equals to the
+// input ones.
+void RemixAndResample(const AudioFrame& src_frame,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
+
+// This version has a pointer to the samples `src_data` as input and receives
+// `samples_per_channel`, `num_channels` and `sample_rate_hz` of the data as
+// parameters.
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      size_t num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_REMIX_RESAMPLE_H_

TMessagesProj/jni/voip/webrtc/audio/utility/audio_frame_operations.cc

@@ -0,0 +1,294 @@
+/*
+ *  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 "audio/utility/audio_frame_operations.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <utility>
+
+#include "common_audio/include/audio_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+
+// 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
+const size_t kMuteFadeFrames = 128;
+const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
+
+}  // namespace
+
+void AudioFrameOperations::Add(const AudioFrame& frame_to_add,
+                               AudioFrame* result_frame) {
+  // Sanity check.
+  RTC_DCHECK(result_frame);
+  RTC_DCHECK_GT(result_frame->num_channels_, 0);
+  RTC_DCHECK_EQ(result_frame->num_channels_, frame_to_add.num_channels_);
+
+  bool no_previous_data = result_frame->muted();
+  if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) {
+    // Special case we have no data to start with.
+    RTC_DCHECK_EQ(result_frame->samples_per_channel_, 0);
+    result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_;
+    no_previous_data = true;
+  }
+
+  if (result_frame->vad_activity_ == AudioFrame::kVadActive ||
+      frame_to_add.vad_activity_ == AudioFrame::kVadActive) {
+    result_frame->vad_activity_ = AudioFrame::kVadActive;
+  } else if (result_frame->vad_activity_ == AudioFrame::kVadUnknown ||
+             frame_to_add.vad_activity_ == AudioFrame::kVadUnknown) {
+    result_frame->vad_activity_ = AudioFrame::kVadUnknown;
+  }
+
+  if (result_frame->speech_type_ != frame_to_add.speech_type_)
+    result_frame->speech_type_ = AudioFrame::kUndefined;
+
+  if (!frame_to_add.muted()) {
+    const int16_t* in_data = frame_to_add.data();
+    int16_t* out_data = result_frame->mutable_data();
+    size_t length =
+        frame_to_add.samples_per_channel_ * frame_to_add.num_channels_;
+    if (no_previous_data) {
+      std::copy(in_data, in_data + length, out_data);
+    } else {
+      for (size_t i = 0; i < length; i++) {
+        const int32_t wrap_guard = static_cast<int32_t>(out_data[i]) +
+                                   static_cast<int32_t>(in_data[i]);
+        out_data[i] = rtc::saturated_cast<int16_t>(wrap_guard);
+      }
+    }
+  }
+}
+
+int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
+  if (frame->num_channels_ != 1) {
+    return -1;
+  }
+  UpmixChannels(2, frame);
+  return 0;
+}
+
+int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
+  if (frame->num_channels_ != 2) {
+    return -1;
+  }
+  DownmixChannels(1, frame);
+  return frame->num_channels_ == 1 ? 0 : -1;
+}
+
+void AudioFrameOperations::QuadToStereo(const int16_t* src_audio,
+                                        size_t samples_per_channel,
+                                        int16_t* dst_audio) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    dst_audio[i * 2] =
+        (static_cast<int32_t>(src_audio[4 * i]) + src_audio[4 * i + 1]) >> 1;
+    dst_audio[i * 2 + 1] =
+        (static_cast<int32_t>(src_audio[4 * i + 2]) + src_audio[4 * i + 3]) >>
+        1;
+  }
+}
+
+int AudioFrameOperations::QuadToStereo(AudioFrame* frame) {
+  if (frame->num_channels_ != 4) {
+    return -1;
+  }
+
+  RTC_DCHECK_LE(frame->samples_per_channel_ * 4,
+                AudioFrame::kMaxDataSizeSamples);
+
+  if (!frame->muted()) {
+    QuadToStereo(frame->data(), frame->samples_per_channel_,
+                 frame->mutable_data());
+  }
+  frame->num_channels_ = 2;
+
+  return 0;
+}
+
+void AudioFrameOperations::DownmixChannels(const int16_t* src_audio,
+                                           size_t src_channels,
+                                           size_t samples_per_channel,
+                                           size_t dst_channels,
+                                           int16_t* dst_audio) {
+  if (src_channels > 1 && dst_channels == 1) {
+    DownmixInterleavedToMono(src_audio, samples_per_channel, src_channels,
+                             dst_audio);
+    return;
+  } else if (src_channels == 4 && dst_channels == 2) {
+    QuadToStereo(src_audio, samples_per_channel, dst_audio);
+    return;
+  }
+
+  RTC_DCHECK_NOTREACHED() << "src_channels: " << src_channels
+                          << ", dst_channels: " << dst_channels;
+}
+
+void AudioFrameOperations::DownmixChannels(size_t dst_channels,
+                                           AudioFrame* frame) {
+  RTC_DCHECK_LE(frame->samples_per_channel_ * frame->num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  if (frame->num_channels_ > 1 && dst_channels == 1) {
+    if (!frame->muted()) {
+      DownmixInterleavedToMono(frame->data(), frame->samples_per_channel_,
+                               frame->num_channels_, frame->mutable_data());
+    }
+    frame->num_channels_ = 1;
+  } else if (frame->num_channels_ == 4 && dst_channels == 2) {
+    int err = QuadToStereo(frame);
+    RTC_DCHECK_EQ(err, 0);
+  } else {
+    RTC_DCHECK_NOTREACHED() << "src_channels: " << frame->num_channels_
+                            << ", dst_channels: " << dst_channels;
+  }
+}
+
+void AudioFrameOperations::UpmixChannels(size_t target_number_of_channels,
+                                         AudioFrame* frame) {
+  RTC_DCHECK_EQ(frame->num_channels_, 1);
+  RTC_DCHECK_LE(frame->samples_per_channel_ * target_number_of_channels,
+                AudioFrame::kMaxDataSizeSamples);
+
+  if (frame->num_channels_ != 1 ||
+      frame->samples_per_channel_ * target_number_of_channels >
+          AudioFrame::kMaxDataSizeSamples) {
+    return;
+  }
+
+  if (!frame->muted()) {
+    // Up-mixing done in place. Going backwards through the frame ensure nothing
+    // is irrevocably overwritten.
+    int16_t* frame_data = frame->mutable_data();
+    for (int i = frame->samples_per_channel_ - 1; i >= 0; i--) {
+      for (size_t j = 0; j < target_number_of_channels; ++j) {
+        frame_data[target_number_of_channels * i + j] = frame_data[i];
+      }
+    }
+  }
+  frame->num_channels_ = target_number_of_channels;
+}
+
+void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  if (frame->num_channels_ != 2 || frame->muted()) {
+    return;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+    std::swap(frame_data[i], frame_data[i + 1]);
+  }
+}
+
+void AudioFrameOperations::Mute(AudioFrame* frame,
+                                bool previous_frame_muted,
+                                bool current_frame_muted) {
+  RTC_DCHECK(frame);
+  if (!previous_frame_muted && !current_frame_muted) {
+    // Not muted, don't touch.
+  } else if (previous_frame_muted && current_frame_muted) {
+    // Frame fully muted.
+    size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
+    RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
+    frame->Mute();
+  } else {
+    // Fade is a no-op on a muted frame.
+    if (frame->muted()) {
+      return;
+    }
+
+    // Limit number of samples to fade, if frame isn't long enough.
+    size_t count = kMuteFadeFrames;
+    float inc = kMuteFadeInc;
+    if (frame->samples_per_channel_ < kMuteFadeFrames) {
+      count = frame->samples_per_channel_;
+      if (count > 0) {
+        inc = 1.0f / count;
+      }
+    }
+
+    size_t start = 0;
+    size_t end = count;
+    float start_g = 0.0f;
+    if (current_frame_muted) {
+      // Fade out the last `count` samples of frame.
+      RTC_DCHECK(!previous_frame_muted);
+      start = frame->samples_per_channel_ - count;
+      end = frame->samples_per_channel_;
+      start_g = 1.0f;
+      inc = -inc;
+    } else {
+      // Fade in the first `count` samples of frame.
+      RTC_DCHECK(previous_frame_muted);
+    }
+
+    // Perform fade.
+    int16_t* frame_data = frame->mutable_data();
+    size_t channels = frame->num_channels_;
+    for (size_t j = 0; j < channels; ++j) {
+      float g = start_g;
+      for (size_t i = start * channels; i < end * channels; i += channels) {
+        g += inc;
+        frame_data[i + j] *= g;
+      }
+    }
+  }
+}
+
+void AudioFrameOperations::Mute(AudioFrame* frame) {
+  Mute(frame, true, true);
+}
+
+void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_GT(frame->num_channels_, 0);
+  if (frame->num_channels_ < 1 || frame->muted()) {
+    return;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = frame_data[i] >> 1;
+  }
+}
+
+int AudioFrameOperations::Scale(float left, float right, AudioFrame* frame) {
+  if (frame->num_channels_ != 2) {
+    return -1;
+  } else if (frame->muted()) {
+    return 0;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame_data[2 * i] = static_cast<int16_t>(left * frame_data[2 * i]);
+    frame_data[2 * i + 1] = static_cast<int16_t>(right * frame_data[2 * i + 1]);
+  }
+  return 0;
+}
+
+int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame* frame) {
+  if (frame->muted()) {
+    return 0;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = rtc::saturated_cast<int16_t>(scale * frame_data[i]);
+  }
+  return 0;
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/utility/audio_frame_operations.h

@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+#define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
+// Change reference parameters to pointers. Consider using a namespace rather
+// than a class.
+class AudioFrameOperations {
+ public:
+  // Add samples in `frame_to_add` with samples in `result_frame`
+  // putting the results in `results_frame`.  The fields
+  // `vad_activity_` and `speech_type_` of the result frame are
+  // updated. If `result_frame` is empty (`samples_per_channel_`==0),
+  // the samples in `frame_to_add` are added to it.  The number of
+  // channels and number of samples per channel must match except when
+  // `result_frame` is empty.
+  static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
+
+  // `frame.num_channels_` will be updated. This version checks for sufficient
+  // buffer size and that `num_channels_` is mono. Use UpmixChannels
+  // instead. TODO(bugs.webrtc.org/8649): remove.
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int MonoToStereo(AudioFrame* frame);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` is stereo. Use DownmixChannels
+  // instead. TODO(bugs.webrtc.org/8649): remove.
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int StereoToMono(AudioFrame* frame);
+
+  // Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place
+  // operation, meaning `src_audio` and `dst_audio` may point to the same
+  // buffer.
+  static void QuadToStereo(const int16_t* src_audio,
+                           size_t samples_per_channel,
+                           int16_t* dst_audio);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` is 4 channels.
+  static int QuadToStereo(AudioFrame* frame);
+
+  // Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`.
+  // This is an in-place operation, meaning `src_audio` and `dst_audio`
+  // may point to the same buffer. Supported channel combinations are
+  // Stereo to Mono, Quad to Mono, and Quad to Stereo.
+  static void DownmixChannels(const int16_t* src_audio,
+                              size_t src_channels,
+                              size_t samples_per_channel,
+                              size_t dst_channels,
+                              int16_t* dst_audio);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` and `dst_channels` are valid and performs relevant downmix.
+  // Supported channel combinations are N channels to Mono, and Quad to Stereo.
+  static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` and `dst_channels` are valid and performs relevant
+  // downmix. Supported channel combinations are Mono to N
+  // channels. The single channel is replicated.
+  static void UpmixChannels(size_t target_number_of_channels,
+                            AudioFrame* frame);
+
+  // Swap the left and right channels of `frame`. Fails silently if `frame` is
+  // not stereo.
+  static void SwapStereoChannels(AudioFrame* frame);
+
+  // Conditionally zero out contents of `frame` for implementing audio mute:
+  //  `previous_frame_muted` &&  `current_frame_muted` - Zero out whole frame.
+  //  `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start.
+  // !`previous_frame_muted` &&  `current_frame_muted` - Fade-out at frame end.
+  // !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched.
+  static void Mute(AudioFrame* frame,
+                   bool previous_frame_muted,
+                   bool current_frame_muted);
+
+  // Zero out contents of frame.
+  static void Mute(AudioFrame* frame);
+
+  // Halve samples in `frame`.
+  static void ApplyHalfGain(AudioFrame* frame);
+
+  static int Scale(float left, float right, AudioFrame* frame);
+
+  static int ScaleWithSat(float scale, AudioFrame* frame);
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_

TMessagesProj/jni/voip/webrtc/audio/utility/channel_mixer.cc

@@ -0,0 +1,99 @@
+/*
+ *  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 "audio/utility/channel_mixer.h"
+
+#include "audio/utility/channel_mixing_matrix.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+ChannelMixer::ChannelMixer(ChannelLayout input_layout,
+                           ChannelLayout output_layout)
+    : input_layout_(input_layout),
+      output_layout_(output_layout),
+      input_channels_(ChannelLayoutToChannelCount(input_layout)),
+      output_channels_(ChannelLayoutToChannelCount(output_layout)) {
+  // Create the transformation matrix.
+  ChannelMixingMatrix matrix_builder(input_layout_, input_channels_,
+                                     output_layout_, output_channels_);
+  remapping_ = matrix_builder.CreateTransformationMatrix(&matrix_);
+}
+
+ChannelMixer::~ChannelMixer() = default;
+
+void ChannelMixer::Transform(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_EQ(matrix_[0].size(), static_cast<size_t>(input_channels_));
+  RTC_DCHECK_EQ(matrix_.size(), static_cast<size_t>(output_channels_));
+
+  // Leave the audio frame intact if the channel layouts for in and out are
+  // identical.
+  if (input_layout_ == output_layout_) {
+    return;
+  }
+
+  if (IsUpMixing()) {
+    RTC_CHECK_LE(frame->samples_per_channel() * output_channels_,
+                 frame->max_16bit_samples());
+  }
+
+  // Only change the number of output channels if the audio frame is muted.
+  if (frame->muted()) {
+    frame->num_channels_ = output_channels_;
+    frame->channel_layout_ = output_layout_;
+    return;
+  }
+
+  const int16_t* in_audio = frame->data();
+
+  // Only allocate fresh memory at first access or if the required size has
+  // increased.
+  // TODO(henrika): we might be able to do downmixing in-place and thereby avoid
+  // extra memory allocation and a memcpy.
+  const size_t num_elements = frame->samples_per_channel() * output_channels_;
+  if (audio_vector_ == nullptr || num_elements > audio_vector_size_) {
+    audio_vector_.reset(new int16_t[num_elements]);
+    audio_vector_size_ = num_elements;
+  }
+  int16_t* out_audio = audio_vector_.get();
+
+  // Modify the number of channels by creating a weighted sum of input samples
+  // where the weights (scale factors) for each output sample are given by the
+  // transformation matrix.
+  for (size_t i = 0; i < frame->samples_per_channel(); i++) {
+    for (size_t output_ch = 0; output_ch < output_channels_; ++output_ch) {
+      float acc_value = 0.0f;
+      for (size_t input_ch = 0; input_ch < input_channels_; ++input_ch) {
+        const float scale = matrix_[output_ch][input_ch];
+        // Scale should always be positive.
+        RTC_DCHECK_GE(scale, 0);
+        // Each output sample is a weighted sum of input samples.
+        acc_value += scale * in_audio[i * input_channels_ + input_ch];
+      }
+      const size_t index = output_channels_ * i + output_ch;
+      RTC_CHECK_LE(index, audio_vector_size_);
+      out_audio[index] = rtc::saturated_cast<int16_t>(acc_value);
+    }
+  }
+
+  // Update channel information.
+  frame->num_channels_ = output_channels_;
+  frame->channel_layout_ = output_layout_;
+
+  // Copy the output result to the audio frame in `frame`.
+  memcpy(
+      frame->mutable_data(), out_audio,
+      sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels());
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/utility/channel_mixer.h

@@ -0,0 +1,86 @@
+/*
+ *  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 AUDIO_UTILITY_CHANNEL_MIXER_H_
+#define AUDIO_UTILITY_CHANNEL_MIXER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio/channel_layout.h"
+
+namespace webrtc {
+
+// ChannelMixer is for converting audio between channel layouts.  The conversion
+// matrix is built upon construction and used during each Transform() call.  The
+// algorithm works by generating a conversion matrix mapping each output channel
+// to list of input channels.  The transform renders all of the output channels,
+// with each output channel rendered according to a weighted sum of the relevant
+// input channels as defined in the matrix.
+// This file is derived from Chromium's media/base/channel_mixer.h.
+class ChannelMixer {
+ public:
+  // To mix two channels into one and preserve loudness, we must apply
+  // (1 / sqrt(2)) gain to each.
+  static constexpr float kHalfPower = 0.707106781186547524401f;
+
+  ChannelMixer(ChannelLayout input_layout, ChannelLayout output_layout);
+  ~ChannelMixer();
+
+  // Transforms all input channels corresponding to the selected `input_layout`
+  // to the number of channels in the selected `output_layout`.
+  // Example usage (downmix from stereo to mono):
+  //
+  //   ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
+  //   AudioFrame frame;
+  //   frame.samples_per_channel_ = 160;
+  //   frame.num_channels_ = 2;
+  //   EXPECT_EQ(2u, frame.channels());
+  //   mixer.Transform(&frame);
+  //   EXPECT_EQ(1u, frame.channels());
+  //
+  void Transform(AudioFrame* frame);
+
+ private:
+  bool IsUpMixing() const { return output_channels_ > input_channels_; }
+
+  // Selected channel layouts.
+  const ChannelLayout input_layout_;
+  const ChannelLayout output_layout_;
+
+  // Channel counts for input and output.
+  const size_t input_channels_;
+  const size_t output_channels_;
+
+  // 2D matrix of output channels to input channels.
+  std::vector<std::vector<float> > matrix_;
+
+  // 1D array used as temporary storage during the transformation.
+  std::unique_ptr<int16_t[]> audio_vector_;
+
+  // Number of elements allocated for `audio_vector_`.
+  size_t audio_vector_size_ = 0;
+
+  // Optimization case for when we can simply remap the input channels to output
+  // channels, i.e., when all scaling factors in `matrix_` equals 1.0.
+  bool remapping_;
+
+  // Delete the copy constructor and assignment operator.
+  ChannelMixer(const ChannelMixer& other) = delete;
+  ChannelMixer& operator=(const ChannelMixer& other) = delete;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_CHANNEL_MIXER_H_

TMessagesProj/jni/voip/webrtc/audio/utility/channel_mixing_matrix.cc

@@ -0,0 +1,333 @@
+/*
+ *  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 "audio/utility/channel_mixing_matrix.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+
+#include "audio/utility/channel_mixer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+// Selects the default usage of VoIP channel mapping adjustments.
+bool UseChannelMappingAdjustmentsByDefault() {
+  return !field_trial::IsEnabled(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch");
+}
+
+}  // namespace
+
+static void ValidateLayout(ChannelLayout layout) {
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_NONE);
+  RTC_CHECK_LE(layout, CHANNEL_LAYOUT_MAX);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_UNSUPPORTED);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_DISCRETE);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC);
+
+  // Verify there's at least one channel.  Should always be true here by virtue
+  // of not being one of the invalid layouts, but lets double check to be sure.
+  int channel_count = ChannelLayoutToChannelCount(layout);
+  RTC_DCHECK_GT(channel_count, 0);
+
+  // If we have more than one channel, verify a symmetric layout for sanity.
+  // The unit test will verify all possible layouts, so this can be a DCHECK.
+  // Symmetry allows simplifying the matrix building code by allowing us to
+  // assume that if one channel of a pair exists, the other will too.
+  if (channel_count > 1) {
+    // Assert that LEFT exists if and only if RIGHT exists, and so on.
+    RTC_DCHECK_EQ(ChannelOrder(layout, LEFT) >= 0,
+                  ChannelOrder(layout, RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, SIDE_LEFT) >= 0,
+                  ChannelOrder(layout, SIDE_RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, BACK_LEFT) >= 0,
+                  ChannelOrder(layout, BACK_RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, LEFT_OF_CENTER) >= 0,
+                  ChannelOrder(layout, RIGHT_OF_CENTER) >= 0);
+  } else {
+    RTC_DCHECK_EQ(layout, CHANNEL_LAYOUT_MONO);
+  }
+}
+
+ChannelMixingMatrix::ChannelMixingMatrix(ChannelLayout input_layout,
+                                         int input_channels,
+                                         ChannelLayout output_layout,
+                                         int output_channels)
+    : use_voip_channel_mapping_adjustments_(
+          UseChannelMappingAdjustmentsByDefault()),
+      input_layout_(input_layout),
+      input_channels_(input_channels),
+      output_layout_(output_layout),
+      output_channels_(output_channels) {
+  // Stereo down mix should never be the output layout.
+  RTC_CHECK_NE(output_layout, CHANNEL_LAYOUT_STEREO_DOWNMIX);
+
+  // Verify that the layouts are supported
+  if (input_layout != CHANNEL_LAYOUT_DISCRETE)
+    ValidateLayout(input_layout);
+  if (output_layout != CHANNEL_LAYOUT_DISCRETE)
+    ValidateLayout(output_layout);
+
+  // Special case for 5.0, 5.1 with back channels when upmixed to 7.0, 7.1,
+  // which should map the back LR to side LR.
+  if (input_layout_ == CHANNEL_LAYOUT_5_0_BACK &&
+      output_layout_ == CHANNEL_LAYOUT_7_0) {
+    input_layout_ = CHANNEL_LAYOUT_5_0;
+  } else if (input_layout_ == CHANNEL_LAYOUT_5_1_BACK &&
+             output_layout_ == CHANNEL_LAYOUT_7_1) {
+    input_layout_ = CHANNEL_LAYOUT_5_1;
+  }
+}
+
+ChannelMixingMatrix::~ChannelMixingMatrix() = default;
+
+bool ChannelMixingMatrix::CreateTransformationMatrix(
+    std::vector<std::vector<float>>* matrix) {
+  matrix_ = matrix;
+
+  // Size out the initial matrix.
+  matrix_->reserve(output_channels_);
+  for (int output_ch = 0; output_ch < output_channels_; ++output_ch)
+    matrix_->push_back(std::vector<float>(input_channels_, 0));
+
+  // First check for discrete case.
+  if (input_layout_ == CHANNEL_LAYOUT_DISCRETE ||
+      output_layout_ == CHANNEL_LAYOUT_DISCRETE) {
+    // If the number of input channels is more than output channels, then
+    // copy as many as we can then drop the remaining input channels.
+    // If the number of input channels is less than output channels, then
+    // copy them all, then zero out the remaining output channels.
+    int passthrough_channels = std::min(input_channels_, output_channels_);
+    for (int i = 0; i < passthrough_channels; ++i)
+      (*matrix_)[i][i] = 1;
+
+    return true;
+  }
+
+  // If specified, use adjusted channel mapping for the VoIP scenario.
+  if (use_voip_channel_mapping_adjustments_ &&
+      input_layout_ == CHANNEL_LAYOUT_MONO &&
+      ChannelLayoutToChannelCount(output_layout_) >= 2) {
+    // Only place the mono input in the front left and right channels.
+    (*matrix_)[0][0] = 1.f;
+    (*matrix_)[1][0] = 1.f;
+
+    for (size_t output_ch = 2; output_ch < matrix_->size(); ++output_ch) {
+      (*matrix_)[output_ch][0] = 0.f;
+    }
+    return true;
+  }
+
+  // Route matching channels and figure out which ones aren't accounted for.
+  for (Channels ch = LEFT; ch < CHANNELS_MAX + 1;
+       ch = static_cast<Channels>(ch + 1)) {
+    int input_ch_index = ChannelOrder(input_layout_, ch);
+    if (input_ch_index < 0)
+      continue;
+
+    int output_ch_index = ChannelOrder(output_layout_, ch);
+    if (output_ch_index < 0) {
+      unaccounted_inputs_.push_back(ch);
+      continue;
+    }
+
+    RTC_DCHECK_LT(static_cast<size_t>(output_ch_index), matrix_->size());
+    RTC_DCHECK_LT(static_cast<size_t>(input_ch_index),
+                  (*matrix_)[output_ch_index].size());
+    (*matrix_)[output_ch_index][input_ch_index] = 1;
+  }
+
+  // If all input channels are accounted for, there's nothing left to do.
+  if (unaccounted_inputs_.empty()) {
+    // Since all output channels map directly to inputs we can optimize.
+    return true;
+  }
+
+  // Mix front LR into center.
+  if (IsUnaccounted(LEFT)) {
+    // When down mixing to mono from stereo, we need to be careful of full scale
+    // stereo mixes.  Scaling by 1 / sqrt(2) here will likely lead to clipping
+    // so we use 1 / 2 instead.
+    float scale =
+        (output_layout_ == CHANNEL_LAYOUT_MONO && input_channels_ == 2)
+            ? 0.5
+            : ChannelMixer::kHalfPower;
+    Mix(LEFT, CENTER, scale);
+    Mix(RIGHT, CENTER, scale);
+  }
+
+  // Mix center into front LR.
+  if (IsUnaccounted(CENTER)) {
+    // When up mixing from mono, just do a copy to front LR.
+    float scale =
+        (input_layout_ == CHANNEL_LAYOUT_MONO) ? 1 : ChannelMixer::kHalfPower;
+    MixWithoutAccounting(CENTER, LEFT, scale);
+    Mix(CENTER, RIGHT, scale);
+  }
+
+  // Mix back LR into: side LR || back center || front LR || front center.
+  if (IsUnaccounted(BACK_LEFT)) {
+    if (HasOutputChannel(SIDE_LEFT)) {
+      // If the input has side LR, mix back LR into side LR, but instead if the
+      // input doesn't have side LR (but output does) copy back LR to side LR.
+      float scale = HasInputChannel(SIDE_LEFT) ? ChannelMixer::kHalfPower : 1;
+      Mix(BACK_LEFT, SIDE_LEFT, scale);
+      Mix(BACK_RIGHT, SIDE_RIGHT, scale);
+    } else if (HasOutputChannel(BACK_CENTER)) {
+      // Mix back LR into back center.
+      Mix(BACK_LEFT, BACK_CENTER, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, BACK_CENTER, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix back LR into front LR.
+      Mix(BACK_LEFT, LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix back LR into front center.
+      Mix(BACK_LEFT, CENTER, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix side LR into: back LR || back center || front LR || front center.
+  if (IsUnaccounted(SIDE_LEFT)) {
+    if (HasOutputChannel(BACK_LEFT)) {
+      // If the input has back LR, mix side LR into back LR, but instead if the
+      // input doesn't have back LR (but output does) copy side LR to back LR.
+      float scale = HasInputChannel(BACK_LEFT) ? ChannelMixer::kHalfPower : 1;
+      Mix(SIDE_LEFT, BACK_LEFT, scale);
+      Mix(SIDE_RIGHT, BACK_RIGHT, scale);
+    } else if (HasOutputChannel(BACK_CENTER)) {
+      // Mix side LR into back center.
+      Mix(SIDE_LEFT, BACK_CENTER, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, BACK_CENTER, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix side LR into front LR.
+      Mix(SIDE_LEFT, LEFT, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix side LR into front center.
+      Mix(SIDE_LEFT, CENTER, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix back center into: back LR || side LR || front LR || front center.
+  if (IsUnaccounted(BACK_CENTER)) {
+    if (HasOutputChannel(BACK_LEFT)) {
+      // Mix back center into back LR.
+      MixWithoutAccounting(BACK_CENTER, BACK_LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, BACK_RIGHT, ChannelMixer::kHalfPower);
+    } else if (HasOutputChannel(SIDE_LEFT)) {
+      // Mix back center into side LR.
+      MixWithoutAccounting(BACK_CENTER, SIDE_LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, SIDE_RIGHT, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix back center into front LR.
+      // TODO(dalecurtis): Not sure about these values?
+      MixWithoutAccounting(BACK_CENTER, LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix back center into front center.
+      // TODO(dalecurtis): Not sure about these values?
+      Mix(BACK_CENTER, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix LR of center into: front LR || front center.
+  if (IsUnaccounted(LEFT_OF_CENTER)) {
+    if (HasOutputChannel(LEFT)) {
+      // Mix LR of center into front LR.
+      Mix(LEFT_OF_CENTER, LEFT, ChannelMixer::kHalfPower);
+      Mix(RIGHT_OF_CENTER, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix LR of center into front center.
+      Mix(LEFT_OF_CENTER, CENTER, ChannelMixer::kHalfPower);
+      Mix(RIGHT_OF_CENTER, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix LFE into: front center || front LR.
+  if (IsUnaccounted(LFE)) {
+    if (!HasOutputChannel(CENTER)) {
+      // Mix LFE into front LR.
+      MixWithoutAccounting(LFE, LEFT, ChannelMixer::kHalfPower);
+      Mix(LFE, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix LFE into front center.
+      Mix(LFE, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // All channels should now be accounted for.
+  RTC_DCHECK(unaccounted_inputs_.empty());
+
+  // See if the output `matrix_` is simply a remapping matrix.  If each input
+  // channel maps to a single output channel we can simply remap.  Doing this
+  // programmatically is less fragile than logic checks on channel mappings.
+  for (int output_ch = 0; output_ch < output_channels_; ++output_ch) {
+    int input_mappings = 0;
+    for (int input_ch = 0; input_ch < input_channels_; ++input_ch) {
+      // We can only remap if each row contains a single scale of 1.  I.e., each
+      // output channel is mapped from a single unscaled input channel.
+      if ((*matrix_)[output_ch][input_ch] != 1 || ++input_mappings > 1)
+        return false;
+    }
+  }
+
+  // If we've gotten here, `matrix_` is simply a remapping.
+  return true;
+}
+
+void ChannelMixingMatrix::AccountFor(Channels ch) {
+  unaccounted_inputs_.erase(
+      std::find(unaccounted_inputs_.begin(), unaccounted_inputs_.end(), ch));
+}
+
+bool ChannelMixingMatrix::IsUnaccounted(Channels ch) const {
+  return std::find(unaccounted_inputs_.begin(), unaccounted_inputs_.end(),
+                   ch) != unaccounted_inputs_.end();
+}
+
+bool ChannelMixingMatrix::HasInputChannel(Channels ch) const {
+  return ChannelOrder(input_layout_, ch) >= 0;
+}
+
+bool ChannelMixingMatrix::HasOutputChannel(Channels ch) const {
+  return ChannelOrder(output_layout_, ch) >= 0;
+}
+
+void ChannelMixingMatrix::Mix(Channels input_ch,
+                              Channels output_ch,
+                              float scale) {
+  MixWithoutAccounting(input_ch, output_ch, scale);
+  AccountFor(input_ch);
+}
+
+void ChannelMixingMatrix::MixWithoutAccounting(Channels input_ch,
+                                               Channels output_ch,
+                                               float scale) {
+  int input_ch_index = ChannelOrder(input_layout_, input_ch);
+  int output_ch_index = ChannelOrder(output_layout_, output_ch);
+
+  RTC_DCHECK(IsUnaccounted(input_ch));
+  RTC_DCHECK_GE(input_ch_index, 0);
+  RTC_DCHECK_GE(output_ch_index, 0);
+
+  RTC_DCHECK_EQ((*matrix_)[output_ch_index][input_ch_index], 0);
+  (*matrix_)[output_ch_index][input_ch_index] = scale;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/utility/channel_mixing_matrix.h

@@ -0,0 +1,76 @@
+/*
+ *  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 AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
+#define AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
+
+#include <vector>
+
+#include "api/audio/channel_layout.h"
+
+namespace webrtc {
+
+class ChannelMixingMatrix {
+ public:
+  ChannelMixingMatrix(ChannelLayout input_layout,
+                      int input_channels,
+                      ChannelLayout output_layout,
+                      int output_channels);
+
+  ~ChannelMixingMatrix();
+
+  // Create the transformation matrix of input channels to output channels.
+  // Updates the empty matrix with the transformation, and returns true
+  // if the transformation is just a remapping of channels (no mixing).
+  // The size of `matrix` is `output_channels` x `input_channels`, i.e., the
+  // number of rows equals the number of output channels and the number of
+  // columns corresponds to the number of input channels.
+  // This file is derived from Chromium's media/base/channel_mixing_matrix.h.
+  bool CreateTransformationMatrix(std::vector<std::vector<float>>* matrix);
+
+ private:
+  const bool use_voip_channel_mapping_adjustments_;
+
+  // Result transformation of input channels to output channels
+  std::vector<std::vector<float>>* matrix_;
+
+  // Input and output channel layout provided during construction.
+  ChannelLayout input_layout_;
+  int input_channels_;
+  ChannelLayout output_layout_;
+  int output_channels_;
+
+  // Helper variable for tracking which inputs are currently unaccounted,
+  // should be empty after construction completes.
+  std::vector<Channels> unaccounted_inputs_;
+
+  // Helper methods for managing unaccounted input channels.
+  void AccountFor(Channels ch);
+  bool IsUnaccounted(Channels ch) const;
+
+  // Helper methods for checking if `ch` exists in either `input_layout_` or
+  // `output_layout_` respectively.
+  bool HasInputChannel(Channels ch) const;
+  bool HasOutputChannel(Channels ch) const;
+
+  // Helper methods for updating `matrix_` with the proper value for
+  // mixing `input_ch` into `output_ch`.  MixWithoutAccounting() does not
+  // remove the channel from `unaccounted_inputs_`.
+  void Mix(Channels input_ch, Channels output_ch, float scale);
+  void MixWithoutAccounting(Channels input_ch, Channels output_ch, float scale);
+
+  // Delete the copy constructor and assignment operator.
+  ChannelMixingMatrix(const ChannelMixingMatrix& other) = delete;
+  ChannelMixingMatrix& operator=(const ChannelMixingMatrix& other) = delete;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_

TMessagesProj/jni/voip/webrtc/audio/voip/audio_channel.cc

@@ -0,0 +1,173 @@
+/*
+ *  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 "audio/voip/audio_channel.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_format.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kRtcpReportIntervalMs = 5000;
+
+}  // namespace
+
+AudioChannel::AudioChannel(
+    Transport* transport,
+    uint32_t local_ssrc,
+    TaskQueueFactory* task_queue_factory,
+    AudioMixer* audio_mixer,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : audio_mixer_(audio_mixer) {
+  RTC_DCHECK(task_queue_factory);
+  RTC_DCHECK(audio_mixer);
+
+  Clock* clock = Clock::GetRealTimeClock();
+  receive_statistics_ = ReceiveStatistics::Create(clock);
+
+  RtpRtcpInterface::Configuration rtp_config;
+  rtp_config.clock = clock;
+  rtp_config.audio = true;
+  rtp_config.receive_statistics = receive_statistics_.get();
+  rtp_config.rtcp_report_interval_ms = kRtcpReportIntervalMs;
+  rtp_config.outgoing_transport = transport;
+  rtp_config.local_media_ssrc = local_ssrc;
+
+  rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
+
+  rtp_rtcp_->SetSendingMediaStatus(false);
+  rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
+
+  ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), clock,
+                                            receive_statistics_.get(),
+                                            std::move(decoder_factory));
+  egress_ =
+      std::make_unique<AudioEgress>(rtp_rtcp_.get(), clock, task_queue_factory);
+
+  // Set the instance of audio ingress to be part of audio mixer for ADM to
+  // fetch audio samples to play.
+  audio_mixer_->AddSource(ingress_.get());
+}
+
+AudioChannel::~AudioChannel() {
+  if (egress_->IsSending()) {
+    StopSend();
+  }
+  if (ingress_->IsPlaying()) {
+    StopPlay();
+  }
+
+  audio_mixer_->RemoveSource(ingress_.get());
+
+  // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `egress_`
+  // here.
+  egress_.reset();
+  ingress_.reset();
+}
+
+bool AudioChannel::StartSend() {
+  // If encoder has not been set, return false.
+  if (!egress_->StartSend()) {
+    return false;
+  }
+
+  // Start sending with RTP stack if it has not been sending yet.
+  if (!rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(true);
+  }
+  return true;
+}
+
+void AudioChannel::StopSend() {
+  egress_->StopSend();
+
+  // Deactivate RTP stack when both sending and receiving are stopped.
+  // SetSendingStatus(false) triggers the transmission of RTCP BYE
+  // message to remote endpoint.
+  if (!ingress_->IsPlaying() && rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(false);
+  }
+}
+
+bool AudioChannel::StartPlay() {
+  // If decoders have not been set, return false.
+  if (!ingress_->StartPlay()) {
+    return false;
+  }
+
+  // If RTP stack is not sending then start sending as in recv-only mode, RTCP
+  // receiver report is expected.
+  if (!rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(true);
+  }
+  return true;
+}
+
+void AudioChannel::StopPlay() {
+  ingress_->StopPlay();
+
+  // Deactivate RTP stack only when both sending and receiving are stopped.
+  if (!rtp_rtcp_->SendingMedia() && rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(false);
+  }
+}
+
+IngressStatistics AudioChannel::GetIngressStatistics() {
+  IngressStatistics ingress_stats;
+  NetworkStatistics stats = ingress_->GetNetworkStatistics();
+  ingress_stats.neteq_stats.total_samples_received = stats.totalSamplesReceived;
+  ingress_stats.neteq_stats.concealed_samples = stats.concealedSamples;
+  ingress_stats.neteq_stats.concealment_events = stats.concealmentEvents;
+  ingress_stats.neteq_stats.jitter_buffer_delay_ms = stats.jitterBufferDelayMs;
+  ingress_stats.neteq_stats.jitter_buffer_emitted_count =
+      stats.jitterBufferEmittedCount;
+  ingress_stats.neteq_stats.jitter_buffer_target_delay_ms =
+      stats.jitterBufferTargetDelayMs;
+  ingress_stats.neteq_stats.inserted_samples_for_deceleration =
+      stats.insertedSamplesForDeceleration;
+  ingress_stats.neteq_stats.removed_samples_for_acceleration =
+      stats.removedSamplesForAcceleration;
+  ingress_stats.neteq_stats.silent_concealed_samples =
+      stats.silentConcealedSamples;
+  ingress_stats.neteq_stats.fec_packets_received = stats.fecPacketsReceived;
+  ingress_stats.neteq_stats.fec_packets_discarded = stats.fecPacketsDiscarded;
+  ingress_stats.neteq_stats.delayed_packet_outage_samples =
+      stats.delayedPacketOutageSamples;
+  ingress_stats.neteq_stats.relative_packet_arrival_delay_ms =
+      stats.relativePacketArrivalDelayMs;
+  ingress_stats.neteq_stats.interruption_count = stats.interruptionCount;
+  ingress_stats.neteq_stats.total_interruption_duration_ms =
+      stats.totalInterruptionDurationMs;
+  ingress_stats.total_duration = ingress_->GetOutputTotalDuration();
+  return ingress_stats;
+}
+
+ChannelStatistics AudioChannel::GetChannelStatistics() {
+  ChannelStatistics channel_stat = ingress_->GetChannelStatistics();
+
+  StreamDataCounters rtp_stats, rtx_stats;
+  rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
+  channel_stat.bytes_sent =
+      rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
+  channel_stat.packets_sent =
+      rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+
+  return channel_stat;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/voip/audio_channel.h

@@ -0,0 +1,131 @@
+/*
+ *  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 AUDIO_VOIP_AUDIO_CHANNEL_H_
+#define AUDIO_VOIP_AUDIO_CHANNEL_H_
+
+#include <map>
+#include <memory>
+#include <queue>
+#include <utility>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_statistics.h"
+#include "audio/voip/audio_egress.h"
+#include "audio/voip/audio_ingress.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// AudioChannel represents a single media session and provides APIs over
+// AudioIngress and AudioEgress. Note that a single RTP stack is shared with
+// these two classes as it has both sending and receiving capabilities.
+class AudioChannel : public rtc::RefCountInterface {
+ public:
+  AudioChannel(Transport* transport,
+               uint32_t local_ssrc,
+               TaskQueueFactory* task_queue_factory,
+               AudioMixer* audio_mixer,
+               rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  ~AudioChannel() override;
+
+  // Set and get ChannelId that this audio channel belongs for debugging and
+  // logging purpose.
+  void SetId(ChannelId id) { id_ = id; }
+  ChannelId GetId() const { return id_; }
+
+  // APIs to start/stop audio channel on each direction.
+  // StartSend/StartPlay returns false if encoder/decoders
+  // have not been set, respectively.
+  bool StartSend();
+  void StopSend();
+  bool StartPlay();
+  void StopPlay();
+
+  // APIs relayed to AudioEgress.
+  bool IsSendingMedia() const { return egress_->IsSending(); }
+  AudioSender* GetAudioSender() { return egress_.get(); }
+  void SetEncoder(int payload_type,
+                  const SdpAudioFormat& encoder_format,
+                  std::unique_ptr<AudioEncoder> encoder) {
+    egress_->SetEncoder(payload_type, encoder_format, std::move(encoder));
+  }
+  absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+    return egress_->GetEncoderFormat();
+  }
+  void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz) {
+    egress_->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz);
+  }
+  bool SendTelephoneEvent(int dtmf_event, int duration_ms) {
+    return egress_->SendTelephoneEvent(dtmf_event, duration_ms);
+  }
+  void SetMute(bool enable) { egress_->SetMute(enable); }
+
+  // APIs relayed to AudioIngress.
+  bool IsPlaying() const { return ingress_->IsPlaying(); }
+  void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
+    ingress_->ReceivedRTPPacket(rtp_packet);
+  }
+  void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet) {
+    ingress_->ReceivedRTCPPacket(rtcp_packet);
+  }
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+    ingress_->SetReceiveCodecs(codecs);
+  }
+  IngressStatistics GetIngressStatistics();
+  ChannelStatistics GetChannelStatistics();
+
+  // See comments on the methods used from AudioEgress and AudioIngress.
+  // Conversion to double is following what is done in
+  // DoubleAudioLevelFromIntAudioLevel method in rtc_stats_collector.cc to be
+  // consistent.
+  double GetInputAudioLevel() const {
+    return egress_->GetInputAudioLevel() / 32767.0;
+  }
+  double GetInputTotalEnergy() const { return egress_->GetInputTotalEnergy(); }
+  double GetInputTotalDuration() const {
+    return egress_->GetInputTotalDuration();
+  }
+  double GetOutputAudioLevel() const {
+    return ingress_->GetOutputAudioLevel() / 32767.0;
+  }
+  double GetOutputTotalEnergy() const {
+    return ingress_->GetOutputTotalEnergy();
+  }
+  double GetOutputTotalDuration() const {
+    return ingress_->GetOutputTotalDuration();
+  }
+
+  // Internal API for testing purpose.
+  void SendRTCPReportForTesting(RTCPPacketType type) {
+    int32_t result = rtp_rtcp_->SendRTCP(type);
+    RTC_DCHECK(result == 0);
+  }
+
+ private:
+  // ChannelId that this audio channel belongs for logging purpose.
+  ChannelId id_;
+
+  // Synchronization is handled internally by AudioMixer.
+  AudioMixer* audio_mixer_;
+
+  // Listed in order for safe destruction of AudioChannel object.
+  // Synchronization for these are handled internally.
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  std::unique_ptr<AudioIngress> ingress_;
+  std::unique_ptr<AudioEgress> egress_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_CHANNEL_H_

TMessagesProj/jni/voip/webrtc/audio/voip/audio_egress.cc

@@ -0,0 +1,190 @@
+/*
+ *  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 "audio/voip/audio_egress.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp,
+                         Clock* clock,
+                         TaskQueueFactory* task_queue_factory)
+    : rtp_rtcp_(rtp_rtcp),
+      rtp_sender_audio_(clock, rtp_rtcp_->RtpSender()),
+      audio_coding_(AudioCodingModule::Create()),
+      encoder_queue_(task_queue_factory->CreateTaskQueue(
+          "AudioEncoder",
+          TaskQueueFactory::Priority::NORMAL)),
+      encoder_queue_checker_(encoder_queue_.get()) {
+  audio_coding_->RegisterTransportCallback(this);
+}
+
+AudioEgress::~AudioEgress() {
+  audio_coding_->RegisterTransportCallback(nullptr);
+
+  // Delete first to ensure that there are no running tasks when the other
+  // members are destroyed.
+  encoder_queue_ = nullptr;
+}
+
+bool AudioEgress::IsSending() const {
+  return rtp_rtcp_->SendingMedia();
+}
+
+void AudioEgress::SetEncoder(int payload_type,
+                             const SdpAudioFormat& encoder_format,
+                             std::unique_ptr<AudioEncoder> encoder) {
+  RTC_DCHECK_GE(payload_type, 0);
+  RTC_DCHECK_LE(payload_type, 127);
+
+  SetEncoderFormat(encoder_format);
+
+  // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
+  // as well as some other things, so we collect this info and send it along.
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type,
+                                          encoder->RtpTimestampRateHz());
+  rtp_sender_audio_.RegisterAudioPayload("audio", payload_type,
+                                         encoder->RtpTimestampRateHz(),
+                                         encoder->NumChannels(), 0);
+
+  audio_coding_->SetEncoder(std::move(encoder));
+}
+
+bool AudioEgress::StartSend() {
+  if (!GetEncoderFormat()) {
+    RTC_DLOG(LS_WARNING) << "Send codec has not been set yet";
+    return false;
+  }
+  rtp_rtcp_->SetSendingMediaStatus(true);
+  return true;
+}
+
+void AudioEgress::StopSend() {
+  rtp_rtcp_->SetSendingMediaStatus(false);
+}
+
+void AudioEgress::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  RTC_DCHECK_LE(audio_frame->num_channels_, 8);
+
+  encoder_queue_->PostTask(
+      [this, audio_frame = std::move(audio_frame)]() mutable {
+        RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+        if (!rtp_rtcp_->SendingMedia()) {
+          return;
+        }
+
+        double duration_seconds =
+            static_cast<double>(audio_frame->samples_per_channel_) /
+            audio_frame->sample_rate_hz_;
+
+        input_audio_level_.ComputeLevel(*audio_frame, duration_seconds);
+
+        AudioFrameOperations::Mute(audio_frame.get(),
+                                   encoder_context_.previously_muted_,
+                                   encoder_context_.mute_);
+        encoder_context_.previously_muted_ = encoder_context_.mute_;
+
+        audio_frame->timestamp_ = encoder_context_.frame_rtp_timestamp_;
+
+        // This call will trigger AudioPacketizationCallback::SendData if
+        // encoding is done and payload is ready for packetization and
+        // transmission. Otherwise, it will return without invoking the
+        // callback.
+        if (audio_coding_->Add10MsData(*audio_frame) < 0) {
+          RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
+          return;
+        }
+
+        encoder_context_.frame_rtp_timestamp_ +=
+            rtc::dchecked_cast<uint32_t>(audio_frame->samples_per_channel_);
+      });
+}
+
+int32_t AudioEgress::SendData(AudioFrameType frame_type,
+                              uint8_t payload_type,
+                              uint32_t timestamp,
+                              const uint8_t* payload_data,
+                              size_t payload_size) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+
+  rtc::ArrayView<const uint8_t> payload(payload_data, payload_size);
+
+  // Currently we don't get a capture time from downstream modules (ADM,
+  // AudioTransportImpl).
+  // TODO(natim@webrtc.org): Integrate once it's ready.
+  constexpr uint32_t kUndefinedCaptureTime = -1;
+
+  // Push data from ACM to RTP/RTCP-module to deliver audio frame for
+  // packetization.
+  if (!rtp_rtcp_->OnSendingRtpFrame(timestamp, kUndefinedCaptureTime,
+                                    payload_type,
+                                    /*force_sender_report=*/false)) {
+    return -1;
+  }
+
+  const uint32_t rtp_timestamp = timestamp + rtp_rtcp_->StartTimestamp();
+
+  // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
+  if (!rtp_sender_audio_.SendAudio({.type = frame_type,
+                                    .payload = payload,
+                                    .payload_id = payload_type,
+                                    .rtp_timestamp = rtp_timestamp})) {
+    RTC_DLOG(LS_ERROR)
+        << "AudioEgress::SendData() failed to send data to RTP/RTCP module";
+    return -1;
+  }
+
+  return 0;
+}
+
+void AudioEgress::RegisterTelephoneEventType(int rtp_payload_type,
+                                             int sample_rate_hz) {
+  RTC_DCHECK_GE(rtp_payload_type, 0);
+  RTC_DCHECK_LE(rtp_payload_type, 127);
+
+  rtp_rtcp_->RegisterSendPayloadFrequency(rtp_payload_type, sample_rate_hz);
+  rtp_sender_audio_.RegisterAudioPayload("telephone-event", rtp_payload_type,
+                                         sample_rate_hz, 0, 0);
+}
+
+bool AudioEgress::SendTelephoneEvent(int dtmf_event, int duration_ms) {
+  RTC_DCHECK_GE(dtmf_event, 0);
+  RTC_DCHECK_LE(dtmf_event, 255);
+  RTC_DCHECK_GE(duration_ms, 0);
+  RTC_DCHECK_LE(duration_ms, 65535);
+
+  if (!IsSending()) {
+    return false;
+  }
+
+  constexpr int kTelephoneEventAttenuationdB = 10;
+
+  if (rtp_sender_audio_.SendTelephoneEvent(dtmf_event, duration_ms,
+                                           kTelephoneEventAttenuationdB) != 0) {
+    RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
+    return false;
+  }
+  return true;
+}
+
+void AudioEgress::SetMute(bool mute) {
+  encoder_queue_->PostTask([this, mute] {
+    RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
+    encoder_context_.mute_ = mute;
+  });
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/voip/audio_egress.h

@@ -0,0 +1,158 @@
+/*
+ *  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 AUDIO_VOIP_AUDIO_EGRESS_H_
+#define AUDIO_VOIP_AUDIO_EGRESS_H_
+
+#include <memory>
+#include <string>
+
+#include "api/audio_codecs/audio_format.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "audio/audio_level.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/audio_sender.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+// AudioEgress receives input samples from AudioDeviceModule via
+// AudioTransportImpl through AudioSender interface. Once it encodes the sample
+// via selected encoder through AudioPacketizationCallback interface, the
+// encoded payload will be packetized by the RTP stack, resulting in ready to
+// send RTP packet to remote endpoint.
+//
+// TaskQueue is used to encode and send RTP asynchrounously as some OS platform
+// uses the same thread for both audio input and output sample deliveries which
+// can affect audio quality.
+//
+// Note that this class is originally based on ChannelSend in
+// audio/channel_send.cc with non-audio related logic trimmed as aimed for
+// smaller footprint.
+class AudioEgress : public AudioSender, public AudioPacketizationCallback {
+ public:
+  AudioEgress(RtpRtcpInterface* rtp_rtcp,
+              Clock* clock,
+              TaskQueueFactory* task_queue_factory);
+  ~AudioEgress() override;
+
+  // Set the encoder format and payload type for AudioCodingModule.
+  // It's possible to change the encoder type during its active usage.
+  // `payload_type` must be the type that is negotiated with peer through
+  // offer/answer.
+  void SetEncoder(int payload_type,
+                  const SdpAudioFormat& encoder_format,
+                  std::unique_ptr<AudioEncoder> encoder);
+
+  // Start or stop sending operation of AudioEgress. This will start/stop
+  // the RTP stack also causes encoder queue thread to start/stop
+  // processing input audio samples. StartSend will return false if
+  // a send codec has not been set.
+  bool StartSend();
+  void StopSend();
+
+  // Query the state of the RTP stack. This returns true if StartSend()
+  // called and false if StopSend() is called.
+  bool IsSending() const;
+
+  // Enable or disable Mute state.
+  void SetMute(bool mute);
+
+  // Retrieve current encoder format info. This returns encoder format set
+  // by SetEncoder() and if encoder is not set, this will return nullopt.
+  absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+    MutexLock lock(&lock_);
+    return encoder_format_;
+  }
+
+  // Register the payload type and sample rate for DTMF (RFC 4733) payload.
+  void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz);
+
+  // Send DTMF named event as specified by
+  // https://tools.ietf.org/html/rfc4733#section-3.2
+  // `duration_ms` specifies the duration of DTMF packets that will be emitted
+  // in place of real RTP packets instead.
+  // This will return true when requested dtmf event is successfully scheduled
+  // otherwise false when the dtmf queue reached maximum of 20 events.
+  bool SendTelephoneEvent(int dtmf_event, int duration_ms);
+
+  // See comments on LevelFullRange, TotalEnergy, TotalDuration from
+  // audio/audio_level.h.
+  int GetInputAudioLevel() const { return input_audio_level_.LevelFullRange(); }
+  double GetInputTotalEnergy() const {
+    return input_audio_level_.TotalEnergy();
+  }
+  double GetInputTotalDuration() const {
+    return input_audio_level_.TotalDuration();
+  }
+
+  // Implementation of AudioSender interface.
+  void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
+
+  // Implementation of AudioPacketizationCallback interface.
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size) override;
+
+ private:
+  void SetEncoderFormat(const SdpAudioFormat& encoder_format) {
+    MutexLock lock(&lock_);
+    encoder_format_ = encoder_format;
+  }
+
+  mutable Mutex lock_;
+
+  // Current encoder format selected by caller.
+  absl::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
+
+  // Synchronization is handled internally by RtpRtcp.
+  RtpRtcpInterface* const rtp_rtcp_;
+
+  // Synchronization is handled internally by RTPSenderAudio.
+  RTPSenderAudio rtp_sender_audio_;
+
+  // Synchronization is handled internally by AudioCodingModule.
+  const std::unique_ptr<AudioCodingModule> audio_coding_;
+
+  // Synchronization is handled internally by voe::AudioLevel.
+  voe::AudioLevel input_audio_level_;
+
+  // Struct that holds all variables used by encoder task queue.
+  struct EncoderContext {
+    // Offset used to mark rtp timestamp in sample rate unit in
+    // newly received audio frame from AudioTransport.
+    uint32_t frame_rtp_timestamp_ = 0;
+
+    // Flag to track mute state from caller. `previously_muted_` is used to
+    // track previous state as part of input to AudioFrameOperations::Mute
+    // to implement fading effect when (un)mute is invoked.
+    bool mute_ = false;
+    bool previously_muted_ = false;
+  };
+
+  EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_checker_);
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_EGRESS_H_

TMessagesProj/jni/voip/webrtc/audio/voip/audio_ingress.cc

@@ -0,0 +1,290 @@
+/*
+ *  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 "audio/voip/audio_ingress.h"
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_format.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+acm2::AcmReceiver::Config CreateAcmConfig(
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
+  acm2::AcmReceiver::Config acm_config;
+  acm_config.neteq_config.enable_muted_state = true;
+  acm_config.decoder_factory = decoder_factory;
+  return acm_config;
+}
+
+}  // namespace
+
+AudioIngress::AudioIngress(
+    RtpRtcpInterface* rtp_rtcp,
+    Clock* clock,
+    ReceiveStatistics* receive_statistics,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : playing_(false),
+      remote_ssrc_(0),
+      first_rtp_timestamp_(-1),
+      rtp_receive_statistics_(receive_statistics),
+      rtp_rtcp_(rtp_rtcp),
+      acm_receiver_(CreateAcmConfig(decoder_factory)),
+      ntp_estimator_(clock) {}
+
+AudioIngress::~AudioIngress() = default;
+
+AudioMixer::Source::AudioFrameInfo AudioIngress::GetAudioFrameWithInfo(
+    int sampling_rate,
+    AudioFrame* audio_frame) {
+  audio_frame->sample_rate_hz_ = sampling_rate;
+
+  // Get 10ms raw PCM data from the ACM.
+  bool muted = false;
+  if (acm_receiver_.GetAudio(sampling_rate, audio_frame, &muted) == -1) {
+    RTC_DLOG(LS_ERROR) << "GetAudio() failed!";
+    // In all likelihood, the audio in this frame is garbage. We return an
+    // error so that the audio mixer module doesn't add it to the mix. As
+    // a result, it won't be played out and the actions skipped here are
+    // irrelevant.
+    return AudioMixer::Source::AudioFrameInfo::kError;
+  }
+
+  if (muted) {
+    AudioFrameOperations::Mute(audio_frame);
+  }
+
+  // Measure audio level.
+  constexpr double kAudioSampleDurationSeconds = 0.01;
+  output_audio_level_.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
+
+  // If caller invoked StopPlay(), then mute the frame.
+  if (!playing_) {
+    AudioFrameOperations::Mute(audio_frame);
+    muted = true;
+  }
+
+  // Set first rtp timestamp with first audio frame with valid timestamp.
+  if (first_rtp_timestamp_ < 0 && audio_frame->timestamp_ != 0) {
+    first_rtp_timestamp_ = audio_frame->timestamp_;
+  }
+
+  if (first_rtp_timestamp_ >= 0) {
+    // Compute elapsed and NTP times.
+    int64_t unwrap_timestamp;
+    {
+      MutexLock lock(&lock_);
+      unwrap_timestamp =
+          timestamp_wrap_handler_.Unwrap(audio_frame->timestamp_);
+      audio_frame->ntp_time_ms_ =
+          ntp_estimator_.Estimate(audio_frame->timestamp_);
+    }
+    // For clock rate, default to the playout sampling rate if we haven't
+    // received any packets yet.
+    absl::optional<std::pair<int, SdpAudioFormat>> decoder =
+        acm_receiver_.LastDecoder();
+    int clock_rate = decoder ? decoder->second.clockrate_hz
+                             : acm_receiver_.last_output_sample_rate_hz();
+    RTC_DCHECK_GT(clock_rate, 0);
+    audio_frame->elapsed_time_ms_ =
+        (unwrap_timestamp - first_rtp_timestamp_) / (clock_rate / 1000);
+  }
+
+  return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
+               : AudioMixer::Source::AudioFrameInfo::kNormal;
+}
+
+bool AudioIngress::StartPlay() {
+  {
+    MutexLock lock(&lock_);
+    if (receive_codec_info_.empty()) {
+      RTC_DLOG(LS_WARNING) << "Receive codecs have not been set yet";
+      return false;
+    }
+  }
+  playing_ = true;
+  return true;
+}
+
+void AudioIngress::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  {
+    MutexLock lock(&lock_);
+    for (const auto& kv : codecs) {
+      receive_codec_info_[kv.first] = kv.second.clockrate_hz;
+    }
+  }
+  acm_receiver_.SetCodecs(codecs);
+}
+
+void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
+  RtpPacketReceived rtp_packet_received;
+  rtp_packet_received.Parse(rtp_packet.data(), rtp_packet.size());
+
+  // Set payload type's sampling rate before we feed it into ReceiveStatistics.
+  {
+    MutexLock lock(&lock_);
+    const auto& it =
+        receive_codec_info_.find(rtp_packet_received.PayloadType());
+    // If sampling rate info is not available in our received codec set, it
+    // would mean that remote media endpoint is sending incorrect payload id
+    // which can't be processed correctly especially on payload type id in
+    // dynamic range.
+    if (it == receive_codec_info_.end()) {
+      RTC_DLOG(LS_WARNING) << "Unexpected payload id received: "
+                           << rtp_packet_received.PayloadType();
+      return;
+    }
+    rtp_packet_received.set_payload_type_frequency(it->second);
+  }
+
+  // Track current remote SSRC.
+  if (rtp_packet_received.Ssrc() != remote_ssrc_) {
+    rtp_rtcp_->SetRemoteSSRC(rtp_packet_received.Ssrc());
+    remote_ssrc_.store(rtp_packet_received.Ssrc());
+  }
+
+  rtp_receive_statistics_->OnRtpPacket(rtp_packet_received);
+
+  RTPHeader header;
+  rtp_packet_received.GetHeader(&header);
+
+  size_t packet_length = rtp_packet_received.size();
+  if (packet_length < header.headerLength ||
+      (packet_length - header.headerLength) < header.paddingLength) {
+    RTC_DLOG(LS_ERROR) << "Packet length(" << packet_length << ") header("
+                       << header.headerLength << ") padding("
+                       << header.paddingLength << ")";
+    return;
+  }
+
+  const uint8_t* payload = rtp_packet_received.data() + header.headerLength;
+  size_t payload_length = packet_length - header.headerLength;
+  size_t payload_data_length = payload_length - header.paddingLength;
+  auto data_view = rtc::ArrayView<const uint8_t>(payload, payload_data_length);
+
+  // Push the incoming payload (parsed and ready for decoding) into the ACM.
+  if (acm_receiver_.InsertPacket(header, data_view) != 0) {
+    RTC_DLOG(LS_ERROR) << "AudioIngress::ReceivedRTPPacket() unable to "
+                          "push data to the ACM";
+  }
+}
+
+void AudioIngress::ReceivedRTCPPacket(
+    rtc::ArrayView<const uint8_t> rtcp_packet) {
+  rtcp::CommonHeader rtcp_header;
+  if (rtcp_header.Parse(rtcp_packet.data(), rtcp_packet.size()) &&
+      (rtcp_header.type() == rtcp::SenderReport::kPacketType ||
+       rtcp_header.type() == rtcp::ReceiverReport::kPacketType)) {
+    RTC_DCHECK_GE(rtcp_packet.size(), 8);
+
+    uint32_t sender_ssrc =
+        ByteReader<uint32_t>::ReadBigEndian(rtcp_packet.data() + 4);
+
+    // If we don't have remote ssrc at this point, it's likely that remote
+    // endpoint is receive-only or it could have restarted the media.
+    if (sender_ssrc != remote_ssrc_) {
+      rtp_rtcp_->SetRemoteSSRC(sender_ssrc);
+      remote_ssrc_.store(sender_ssrc);
+    }
+  }
+
+  // Deliver RTCP packet to RTP/RTCP module for parsing and processing.
+  rtp_rtcp_->IncomingRtcpPacket(rtcp_packet);
+
+  absl::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
+  if (!rtt.has_value()) {
+    // Waiting for valid RTT.
+    return;
+  }
+
+  absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+      rtp_rtcp_->GetSenderReportStats();
+  if (!last_sr.has_value()) {
+    // Waiting for RTCP.
+    return;
+  }
+
+  {
+    MutexLock lock(&lock_);
+    ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
+                                       last_sr->last_remote_rtp_timestamp);
+  }
+}
+
+ChannelStatistics AudioIngress::GetChannelStatistics() {
+  ChannelStatistics channel_stats;
+
+  // Get clockrate for current decoder ahead of jitter calculation.
+  uint32_t clockrate_hz = 0;
+  absl::optional<std::pair<int, SdpAudioFormat>> decoder =
+      acm_receiver_.LastDecoder();
+  if (decoder) {
+    clockrate_hz = decoder->second.clockrate_hz;
+  }
+
+  StreamStatistician* statistician =
+      rtp_receive_statistics_->GetStatistician(remote_ssrc_);
+  if (statistician) {
+    RtpReceiveStats stats = statistician->GetStats();
+    channel_stats.packets_lost = stats.packets_lost;
+    channel_stats.packets_received = stats.packet_counter.packets;
+    channel_stats.bytes_received = stats.packet_counter.payload_bytes;
+    channel_stats.remote_ssrc = remote_ssrc_;
+    if (clockrate_hz > 0) {
+      channel_stats.jitter = static_cast<double>(stats.jitter) / clockrate_hz;
+    }
+  }
+
+  // Get RTCP report using remote SSRC.
+  const std::vector<ReportBlockData>& report_data =
+      rtp_rtcp_->GetLatestReportBlockData();
+  for (const ReportBlockData& rtcp_report : report_data) {
+    if (rtp_rtcp_->SSRC() != rtcp_report.source_ssrc() ||
+        remote_ssrc_ != rtcp_report.sender_ssrc()) {
+      continue;
+    }
+    RemoteRtcpStatistics remote_stat;
+    remote_stat.packets_lost = rtcp_report.cumulative_lost();
+    remote_stat.fraction_lost = rtcp_report.fraction_lost();
+    if (clockrate_hz > 0) {
+      remote_stat.jitter = rtcp_report.jitter(clockrate_hz).seconds<double>();
+    }
+    if (rtcp_report.has_rtt()) {
+      remote_stat.round_trip_time = rtcp_report.last_rtt().seconds<double>();
+    }
+    remote_stat.last_report_received_timestamp_ms =
+        rtcp_report.report_block_timestamp_utc().ms();
+    channel_stats.remote_rtcp = remote_stat;
+
+    // Receive only channel won't send any RTP packets.
+    if (!channel_stats.remote_ssrc.has_value()) {
+      channel_stats.remote_ssrc = remote_ssrc_;
+    }
+    break;
+  }
+
+  return channel_stats;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/voip/audio_ingress.h

@@ -0,0 +1,145 @@
+/*
+ *  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 AUDIO_VOIP_AUDIO_INGRESS_H_
+#define AUDIO_VOIP_AUDIO_INGRESS_H_
+
+#include <algorithm>
+#include <atomic>
+#include <map>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/audio_mixer.h"
+#include "api/rtp_headers.h"
+#include "api/scoped_refptr.h"
+#include "api/voip/voip_statistics.h"
+#include "audio/audio_level.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// AudioIngress handles incoming RTP/RTCP packets from the remote
+// media endpoint. Received RTP packets are injected into AcmReceiver and
+// when audio output thread requests for audio samples to play through system
+// output such as speaker device, AudioIngress provides the samples via its
+// implementation on AudioMixer::Source interface.
+//
+// Note that this class is originally based on ChannelReceive in
+// audio/channel_receive.cc with non-audio related logic trimmed as aimed for
+// smaller footprint.
+class AudioIngress : public AudioMixer::Source {
+ public:
+  AudioIngress(RtpRtcpInterface* rtp_rtcp,
+               Clock* clock,
+               ReceiveStatistics* receive_statistics,
+               rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  ~AudioIngress() override;
+
+  // Start or stop receiving operation of AudioIngress.
+  bool StartPlay();
+  void StopPlay() {
+    playing_ = false;
+    output_audio_level_.ResetLevelFullRange();
+  }
+
+  // Query the state of the AudioIngress.
+  bool IsPlaying() const { return playing_; }
+
+  // Set the decoder formats and payload type for AcmReceiver where the
+  // key type (int) of the map is the payload type of SdpAudioFormat.
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
+  // APIs to handle received RTP/RTCP packets from caller.
+  void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet);
+  void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet);
+
+  // See comments on LevelFullRange, TotalEnergy, TotalDuration from
+  // audio/audio_level.h.
+  int GetOutputAudioLevel() const {
+    return output_audio_level_.LevelFullRange();
+  }
+  double GetOutputTotalEnergy() { return output_audio_level_.TotalEnergy(); }
+  double GetOutputTotalDuration() {
+    return output_audio_level_.TotalDuration();
+  }
+
+  NetworkStatistics GetNetworkStatistics() const {
+    NetworkStatistics stats;
+    acm_receiver_.GetNetworkStatistics(&stats,
+                                       /*get_and_clear_legacy_stats=*/false);
+    return stats;
+  }
+
+  ChannelStatistics GetChannelStatistics();
+
+  // Implementation of AudioMixer::Source interface.
+  AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sampling_rate,
+      AudioFrame* audio_frame) override;
+  int Ssrc() const override {
+    return rtc::dchecked_cast<int>(remote_ssrc_.load());
+  }
+  int PreferredSampleRate() const override {
+    // If we haven't received any RTP packet from remote and thus
+    // last_packet_sampling_rate is not available then use NetEq's sampling
+    // rate as that would be what would be used for audio output sample.
+    return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
+                    acm_receiver_.last_output_sample_rate_hz());
+  }
+
+ private:
+  // Indicates AudioIngress status as caller invokes Start/StopPlaying.
+  // If not playing, incoming RTP data processing is skipped, thus
+  // producing no data to output device.
+  std::atomic<bool> playing_;
+
+  // Currently active remote ssrc from remote media endpoint.
+  std::atomic<uint32_t> remote_ssrc_;
+
+  // The first rtp timestamp of the output audio frame that is used to
+  // calculate elasped time for subsequent audio frames.
+  std::atomic<int64_t> first_rtp_timestamp_;
+
+  // Synchronizaton is handled internally by ReceiveStatistics.
+  ReceiveStatistics* const rtp_receive_statistics_;
+
+  // Synchronizaton is handled internally by RtpRtcpInterface.
+  RtpRtcpInterface* const rtp_rtcp_;
+
+  // Synchronizaton is handled internally by acm2::AcmReceiver.
+  acm2::AcmReceiver acm_receiver_;
+
+  // Synchronizaton is handled internally by voe::AudioLevel.
+  voe::AudioLevel output_audio_level_;
+
+  Mutex lock_;
+
+  RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(lock_);
+
+  // For receiving RTP statistics, this tracks the sampling rate value
+  // per payload type set when caller set via SetReceiveCodecs.
+  std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
+
+  RtpTimestampUnwrapper timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_INGRESS_H_

TMessagesProj/jni/voip/webrtc/audio/voip/voip_core.cc

@@ -0,0 +1,500 @@
+/*
+ *  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 "audio/voip/voip_core.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// For Windows, use specific enum type to initialize default audio device as
+// defined in AudioDeviceModule::WindowsDeviceType.
+#if defined(WEBRTC_WIN)
+constexpr AudioDeviceModule::WindowsDeviceType kAudioDeviceId =
+    AudioDeviceModule::WindowsDeviceType::kDefaultCommunicationDevice;
+#else
+constexpr uint16_t kAudioDeviceId = 0;
+#endif  // defined(WEBRTC_WIN)
+
+// Maximum value range limit on ChannelId. This can be increased without any
+// side effect and only set at this moderate value for better readability for
+// logging.
+static constexpr int kMaxChannelId = 100000;
+
+}  // namespace
+
+VoipCore::VoipCore(rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+                   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+                   std::unique_ptr<TaskQueueFactory> task_queue_factory,
+                   rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
+                   rtc::scoped_refptr<AudioProcessing> audio_processing) {
+  encoder_factory_ = std::move(encoder_factory);
+  decoder_factory_ = std::move(decoder_factory);
+  task_queue_factory_ = std::move(task_queue_factory);
+  audio_device_module_ = std::move(audio_device_module);
+  audio_processing_ = std::move(audio_processing);
+  audio_mixer_ = AudioMixerImpl::Create();
+
+  // AudioTransportImpl depends on audio mixer and audio processing instances.
+  audio_transport_ = std::make_unique<AudioTransportImpl>(
+      audio_mixer_.get(), audio_processing_.get(), nullptr);
+}
+
+bool VoipCore::InitializeIfNeeded() {
+  // `audio_device_module_` internally owns a lock and the whole logic here
+  // needs to be executed atomically once using another lock in VoipCore.
+  // Further changes in this method will need to make sure that no deadlock is
+  // introduced in the future.
+  MutexLock lock(&lock_);
+
+  if (initialized_) {
+    return true;
+  }
+
+  // Initialize ADM.
+  if (audio_device_module_->Init() != 0) {
+    RTC_LOG(LS_ERROR) << "Failed to initialize the ADM.";
+    return false;
+  }
+
+  // Note that failures on initializing default recording/speaker devices are
+  // not considered to be fatal here. In certain case, caller may not care about
+  // recording device functioning (e.g webinar where only speaker is available).
+  // It's also possible that there are other audio devices available that may
+  // work.
+
+  // Initialize default speaker device.
+  if (audio_device_module_->SetPlayoutDevice(kAudioDeviceId) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set playout device.";
+  }
+  if (audio_device_module_->InitSpeaker() != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to access speaker.";
+  }
+
+  // Initialize default recording device.
+  if (audio_device_module_->SetRecordingDevice(kAudioDeviceId) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set recording device.";
+  }
+  if (audio_device_module_->InitMicrophone() != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to access microphone.";
+  }
+
+  // Set number of channels on speaker device.
+  bool available = false;
+  if (audio_device_module_->StereoPlayoutIsAvailable(&available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to query stereo playout.";
+  }
+  if (audio_device_module_->SetStereoPlayout(available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set mono/stereo playout mode.";
+  }
+
+  // Set number of channels on recording device.
+  available = false;
+  if (audio_device_module_->StereoRecordingIsAvailable(&available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to query stereo recording.";
+  }
+  if (audio_device_module_->SetStereoRecording(available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set stereo recording mode.";
+  }
+
+  if (audio_device_module_->RegisterAudioCallback(audio_transport_.get()) !=
+      0) {
+    RTC_LOG(LS_WARNING) << "Unable to register audio callback.";
+  }
+
+  initialized_ = true;
+
+  return true;
+}
+
+ChannelId VoipCore::CreateChannel(Transport* transport,
+                                  absl::optional<uint32_t> local_ssrc) {
+  ChannelId channel_id;
+
+  // Set local ssrc to random if not set by caller.
+  if (!local_ssrc) {
+    Random random(rtc::TimeMicros());
+    local_ssrc = random.Rand<uint32_t>();
+  }
+
+  rtc::scoped_refptr<AudioChannel> channel =
+      rtc::make_ref_counted<AudioChannel>(transport, local_ssrc.value(),
+                                          task_queue_factory_.get(),
+                                          audio_mixer_.get(), decoder_factory_);
+
+  {
+    MutexLock lock(&lock_);
+
+    channel_id = static_cast<ChannelId>(next_channel_id_);
+    channels_[channel_id] = channel;
+    next_channel_id_++;
+    if (next_channel_id_ >= kMaxChannelId) {
+      next_channel_id_ = 0;
+    }
+  }
+
+  // Set ChannelId in audio channel for logging/debugging purpose.
+  channel->SetId(channel_id);
+
+  return channel_id;
+}
+
+VoipResult VoipCore::ReleaseChannel(ChannelId channel_id) {
+  // Destroy channel outside of the lock.
+  rtc::scoped_refptr<AudioChannel> channel;
+
+  bool no_channels_after_release = false;
+
+  {
+    MutexLock lock(&lock_);
+
+    auto iter = channels_.find(channel_id);
+    if (iter != channels_.end()) {
+      channel = std::move(iter->second);
+      channels_.erase(iter);
+    }
+
+    no_channels_after_release = channels_.empty();
+  }
+
+  VoipResult status_code = VoipResult::kOk;
+  if (!channel) {
+    RTC_LOG(LS_WARNING) << "Channel " << channel_id << " not found";
+    status_code = VoipResult::kInvalidArgument;
+  }
+
+  if (no_channels_after_release) {
+    // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `channel`
+    // here.
+    channel = nullptr;
+
+    // Make sure to stop playout on ADM if it is playing.
+    if (audio_device_module_->Playing()) {
+      if (audio_device_module_->StopPlayout() != 0) {
+        RTC_LOG(LS_WARNING) << "StopPlayout failed";
+        status_code = VoipResult::kInternal;
+      }
+    }
+  }
+
+  return status_code;
+}
+
+rtc::scoped_refptr<AudioChannel> VoipCore::GetChannel(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel;
+  {
+    MutexLock lock(&lock_);
+    auto iter = channels_.find(channel_id);
+    if (iter != channels_.end()) {
+      channel = iter->second;
+    }
+  }
+  if (!channel) {
+    RTC_LOG(LS_ERROR) << "Channel " << channel_id << " not found";
+  }
+  return channel;
+}
+
+bool VoipCore::UpdateAudioTransportWithSenders() {
+  std::vector<AudioSender*> audio_senders;
+
+  // Gather a list of audio channel that are currently sending along with
+  // highest sampling rate and channel numbers to configure into audio
+  // transport.
+  int max_sampling_rate = 8000;
+  size_t max_num_channels = 1;
+  {
+    MutexLock lock(&lock_);
+    // Reserve to prevent run time vector re-allocation.
+    audio_senders.reserve(channels_.size());
+    for (auto kv : channels_) {
+      rtc::scoped_refptr<AudioChannel>& channel = kv.second;
+      if (channel->IsSendingMedia()) {
+        auto encoder_format = channel->GetEncoderFormat();
+        if (!encoder_format) {
+          RTC_LOG(LS_ERROR)
+              << "channel " << channel->GetId() << " encoder is not set";
+          continue;
+        }
+        audio_senders.push_back(channel->GetAudioSender());
+        max_sampling_rate =
+            std::max(max_sampling_rate, encoder_format->clockrate_hz);
+        max_num_channels =
+            std::max(max_num_channels, encoder_format->num_channels);
+      }
+    }
+  }
+
+  audio_transport_->UpdateAudioSenders(audio_senders, max_sampling_rate,
+                                       max_num_channels);
+
+  // Depending on availability of senders, turn on or off ADM recording.
+  if (!audio_senders.empty()) {
+    // Initialize audio device module and default device if needed.
+    if (!InitializeIfNeeded()) {
+      return false;
+    }
+
+    if (!audio_device_module_->Recording()) {
+      if (audio_device_module_->InitRecording() != 0) {
+        RTC_LOG(LS_ERROR) << "InitRecording failed";
+        return false;
+      }
+      if (audio_device_module_->StartRecording() != 0) {
+        RTC_LOG(LS_ERROR) << "StartRecording failed";
+        return false;
+      }
+    }
+  } else {
+    if (audio_device_module_->Recording() &&
+        audio_device_module_->StopRecording() != 0) {
+      RTC_LOG(LS_ERROR) << "StopRecording failed";
+      return false;
+    }
+  }
+  return true;
+}
+
+VoipResult VoipCore::StartSend(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  if (!channel->StartSend()) {
+    return VoipResult::kFailedPrecondition;
+  }
+
+  return UpdateAudioTransportWithSenders() ? VoipResult::kOk
+                                           : VoipResult::kInternal;
+}
+
+VoipResult VoipCore::StopSend(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->StopSend();
+
+  return UpdateAudioTransportWithSenders() ? VoipResult::kOk
+                                           : VoipResult::kInternal;
+}
+
+VoipResult VoipCore::StartPlayout(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  if (channel->IsPlaying()) {
+    return VoipResult::kOk;
+  }
+
+  if (!channel->StartPlay()) {
+    return VoipResult::kFailedPrecondition;
+  }
+
+  // Initialize audio device module and default device if needed.
+  if (!InitializeIfNeeded()) {
+    return VoipResult::kInternal;
+  }
+
+  if (!audio_device_module_->Playing()) {
+    if (audio_device_module_->InitPlayout() != 0) {
+      RTC_LOG(LS_ERROR) << "InitPlayout failed";
+      return VoipResult::kInternal;
+    }
+    if (audio_device_module_->StartPlayout() != 0) {
+      RTC_LOG(LS_ERROR) << "StartPlayout failed";
+      return VoipResult::kInternal;
+    }
+  }
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::StopPlayout(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->StopPlay();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::ReceivedRTPPacket(
+    ChannelId channel_id,
+    rtc::ArrayView<const uint8_t> rtp_packet) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->ReceivedRTPPacket(rtp_packet);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::ReceivedRTCPPacket(
+    ChannelId channel_id,
+    rtc::ArrayView<const uint8_t> rtcp_packet) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->ReceivedRTCPPacket(rtcp_packet);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SetSendCodec(ChannelId channel_id,
+                                  int payload_type,
+                                  const SdpAudioFormat& encoder_format) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  auto encoder = encoder_factory_->MakeAudioEncoder(
+      payload_type, encoder_format, absl::nullopt);
+  channel->SetEncoder(payload_type, encoder_format, std::move(encoder));
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SetReceiveCodecs(
+    ChannelId channel_id,
+    const std::map<int, SdpAudioFormat>& decoder_specs) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->SetReceiveCodecs(decoder_specs);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::RegisterTelephoneEventType(ChannelId channel_id,
+                                                int rtp_payload_type,
+                                                int sample_rate_hz) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SendDtmfEvent(ChannelId channel_id,
+                                   DtmfEvent dtmf_event,
+                                   int duration_ms) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  return (channel->SendTelephoneEvent(static_cast<int>(dtmf_event), duration_ms)
+              ? VoipResult::kOk
+              : VoipResult::kFailedPrecondition);
+}
+
+VoipResult VoipCore::GetIngressStatistics(ChannelId channel_id,
+                                          IngressStatistics& ingress_stats) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  ingress_stats = channel->GetIngressStatistics();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::GetChannelStatistics(ChannelId channel_id,
+                                          ChannelStatistics& channel_stats) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel_stats = channel->GetChannelStatistics();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SetInputMuted(ChannelId channel_id, bool enable) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->SetMute(enable);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::GetInputVolumeInfo(ChannelId channel_id,
+                                        VolumeInfo& input_volume) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  input_volume.audio_level = channel->GetInputAudioLevel();
+  input_volume.total_energy = channel->GetInputTotalEnergy();
+  input_volume.total_duration = channel->GetInputTotalDuration();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::GetOutputVolumeInfo(ChannelId channel_id,
+                                         VolumeInfo& output_volume) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  output_volume.audio_level = channel->GetOutputAudioLevel();
+  output_volume.total_energy = channel->GetOutputTotalEnergy();
+  output_volume.total_duration = channel->GetOutputTotalDuration();
+
+  return VoipResult::kOk;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/audio/voip/voip_core.h

@@ -0,0 +1,174 @@
+/*
+ *  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 AUDIO_VOIP_VOIP_CORE_H_
+#define AUDIO_VOIP_VOIP_CORE_H_
+
+#include <map>
+#include <memory>
+#include <queue>
+#include <unordered_map>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_codec.h"
+#include "api/voip/voip_dtmf.h"
+#include "api/voip/voip_engine.h"
+#include "api/voip/voip_network.h"
+#include "api/voip/voip_statistics.h"
+#include "api/voip/voip_volume_control.h"
+#include "audio/audio_transport_impl.h"
+#include "audio/voip/audio_channel.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// VoipCore is the implementatino of VoIP APIs listed in api/voip directory.
+// It manages a vector of AudioChannel objects where each is mapped with a
+// ChannelId (int) type. ChannelId is the primary key to locate a specific
+// AudioChannel object to operate requested VoIP API from the caller.
+//
+// This class receives required audio components from caller at construction and
+// owns the life cycle of them to orchestrate the proper destruction sequence.
+class VoipCore : public VoipEngine,
+                 public VoipBase,
+                 public VoipNetwork,
+                 public VoipCodec,
+                 public VoipDtmf,
+                 public VoipStatistics,
+                 public VoipVolumeControl {
+ public:
+  // Construct VoipCore with provided arguments.
+  VoipCore(rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+           rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+           std::unique_ptr<TaskQueueFactory> task_queue_factory,
+           rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
+           rtc::scoped_refptr<AudioProcessing> audio_processing);
+  ~VoipCore() override = default;
+
+  // Implements VoipEngine interfaces.
+  VoipBase& Base() override { return *this; }
+  VoipNetwork& Network() override { return *this; }
+  VoipCodec& Codec() override { return *this; }
+  VoipDtmf& Dtmf() override { return *this; }
+  VoipStatistics& Statistics() override { return *this; }
+  VoipVolumeControl& VolumeControl() override { return *this; }
+
+  // Implements VoipBase interfaces.
+  ChannelId CreateChannel(Transport* transport,
+                          absl::optional<uint32_t> local_ssrc) override;
+  VoipResult ReleaseChannel(ChannelId channel_id) override;
+  VoipResult StartSend(ChannelId channel_id) override;
+  VoipResult StopSend(ChannelId channel_id) override;
+  VoipResult StartPlayout(ChannelId channel_id) override;
+  VoipResult StopPlayout(ChannelId channel_id) override;
+
+  // Implements VoipNetwork interfaces.
+  VoipResult ReceivedRTPPacket(
+      ChannelId channel_id,
+      rtc::ArrayView<const uint8_t> rtp_packet) override;
+  VoipResult ReceivedRTCPPacket(
+      ChannelId channel_id,
+      rtc::ArrayView<const uint8_t> rtcp_packet) override;
+
+  // Implements VoipCodec interfaces.
+  VoipResult SetSendCodec(ChannelId channel_id,
+                          int payload_type,
+                          const SdpAudioFormat& encoder_format) override;
+  VoipResult SetReceiveCodecs(
+      ChannelId channel_id,
+      const std::map<int, SdpAudioFormat>& decoder_specs) override;
+
+  // Implements VoipDtmf interfaces.
+  VoipResult RegisterTelephoneEventType(ChannelId channel_id,
+                                        int rtp_payload_type,
+                                        int sample_rate_hz) override;
+  VoipResult SendDtmfEvent(ChannelId channel_id,
+                           DtmfEvent dtmf_event,
+                           int duration_ms) override;
+
+  // Implements VoipStatistics interfaces.
+  VoipResult GetIngressStatistics(ChannelId channel_id,
+                                  IngressStatistics& ingress_stats) override;
+  VoipResult GetChannelStatistics(ChannelId channe_id,
+                                  ChannelStatistics& channel_stats) override;
+
+  // Implements VoipVolumeControl interfaces.
+  VoipResult SetInputMuted(ChannelId channel_id, bool enable) override;
+  VoipResult GetInputVolumeInfo(ChannelId channel_id,
+                                VolumeInfo& volume_info) override;
+  VoipResult GetOutputVolumeInfo(ChannelId channel_id,
+                                 VolumeInfo& volume_info) override;
+
+ private:
+  // Initialize ADM and default audio device if needed.
+  // Returns true if ADM is successfully initialized or already in such state
+  // (e.g called more than once). Returns false when ADM fails to initialize
+  // which would presumably render further processing useless. Note that such
+  // failure won't necessarily succeed in next initialization attempt as it
+  // would mean changing the ADM implementation. From Android N and onwards, the
+  // mobile app may not be able to gain microphone access when in background
+  // mode. Therefore it would be better to delay the logic as late as possible.
+  bool InitializeIfNeeded();
+
+  // Fetches the corresponding AudioChannel assigned with given `channel`.
+  // Returns nullptr if not found.
+  rtc::scoped_refptr<AudioChannel> GetChannel(ChannelId channel_id);
+
+  // Updates AudioTransportImpl with a new set of actively sending AudioSender
+  // (AudioEgress). This needs to be invoked whenever StartSend/StopSend is
+  // involved by caller. Returns false when the selected audio device fails to
+  // initialize where it can't expect to deliver any audio input sample.
+  bool UpdateAudioTransportWithSenders();
+
+  // Synchronization for these are handled internally.
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+
+  // Synchronization is handled internally by AudioProcessing.
+  // Must be placed before `audio_device_module_` for proper destruction.
+  rtc::scoped_refptr<AudioProcessing> audio_processing_;
+
+  // Synchronization is handled internally by AudioMixer.
+  // Must be placed before `audio_device_module_` for proper destruction.
+  rtc::scoped_refptr<AudioMixer> audio_mixer_;
+
+  // Synchronization is handled internally by AudioTransportImpl.
+  // Must be placed before `audio_device_module_` for proper destruction.
+  std::unique_ptr<AudioTransportImpl> audio_transport_;
+
+  // Synchronization is handled internally by AudioDeviceModule.
+  rtc::scoped_refptr<AudioDeviceModule> audio_device_module_;
+
+  Mutex lock_;
+
+  // Member to track a next ChannelId for new AudioChannel.
+  int next_channel_id_ RTC_GUARDED_BY(lock_) = 0;
+
+  // Container to track currently active AudioChannel objects mapped by
+  // ChannelId.
+  std::unordered_map<ChannelId, rtc::scoped_refptr<AudioChannel>> channels_
+      RTC_GUARDED_BY(lock_);
+
+  // Boolean flag to ensure initialization only occurs once.
+  bool initialized_ RTC_GUARDED_BY(lock_) = false;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_VOIP_CORE_H_