Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 922 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/channel_send.cc b/third_party/libwebrtc/audio/channel_send.cc
new file mode 100644
index 0000000000..ee94760b6f
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send.cc
@@ -0,0 +1,922 @@
+/*
+ *  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/synchronization/mutex.h"
+#include "rtc_base/task_queue.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 RtcpCounterObserver : public RtcpPacketTypeCounterObserver {
+ public:
+  explicit RtcpCounterObserver(uint32_t ssrc) : ssrc_(ssrc) {}
+
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) override {
+    if (ssrc_ != ssrc) {
+      return;
+    }
+
+    MutexLock lock(&mutex_);
+    packet_counter_ = packet_counter;
+  }
+
+  RtcpPacketTypeCounter GetCounts() {
+    MutexLock lock(&mutex_);
+    return packet_counter_;
+  }
+
+ private:
+  Mutex mutex_;
+  const uint32_t ssrc_;
+  RtcpPacketTypeCounter packet_counter_;
+};
+
+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,
+                  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_RUN_ON(encoder_queue_);
+
+  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.
+  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_);
+  bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false;
+  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_) = false;
+
+  const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_;
+
+  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_;
+
+  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_);
+  // 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_);
+
+  mutable Mutex rtcp_counter_mutex_;
+  RtcpPacketTypeCounter rtcp_packet_type_counter_
+      RTC_GUARDED_BY(rtcp_counter_mutex_);
+
+  // Defined last to ensure that there are no running tasks when the other
+  // members are destroyed.
+  rtc::TaskQueue encoder_queue_;
+};
+
+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_);
+    if (rtp_packet_pacer_) {
+      rtp_packet_pacer_->EnqueuePackets(std::move(packets));
+    }
+  }
+
+  void RemovePacketsForSsrc(uint32_t ssrc) override {
+    MutexLock lock(&mutex_);
+    if (rtp_packet_pacer_) {
+      rtp_packet_pacer_->RemovePacketsForSsrc(ssrc);
+    }
+  }
+
+ private:
+  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_);
+  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.
+    frame_transformer_delegate_->Transform(
+        frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
+        payloadData, payloadSize, absolute_capture_timestamp_ms,
+        rtp_rtcp_->SSRC());
+    return 0;
+  }
+  return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
+                      absolute_capture_timestamp_ms);
+}
+
+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) {
+  // 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()};
+  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),
+      rtcp_counter_observer_(new RtcpCounterObserver(ssrc)),
+      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)) {
+  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;
+  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;
+  configuration.rtcp_packet_type_counter_observer =
+      rtcp_counter_observer_.get();
+  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);
+}
+
+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);
+}
+
+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_);
+    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,
+                             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);
+
+  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(webrtc::AudioLevel::Uri(), id);
+  } else {
+    rtp_rtcp_->DeregisterSendRtpHeaderExtension(webrtc::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();
+  stats.rtcp_packet_type_counts = rtcp_counter_observer_->GetCounts();
+
+  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_);
+        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_);
+    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_);
+        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_);
+  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_DCHECK_RUN_ON(&encoder_queue_);
+        return SendRtpAudio(
+            frameType, payloadType,
+            rtp_timestamp_with_offset - rtp_rtcp_->StartTimestamp(), payload,
+            absolute_capture_timestamp_ms);
+      };
+  frame_transformer_delegate_ =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          std::move(send_audio_callback), std::move(frame_transformer),
+          &encoder_queue_);
+  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