/*
 *  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 VoERtcpObserver;

class ChannelSend : public ChannelSendInterface,
                    public AudioPacketizationCallback,  // receive encoded
                                                        // packets from the ACM
                    public RtcpPacketTypeCounterObserver {
 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,
              TransportFeedbackObserver* feedback_observer,
              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,
      RtcpBandwidthObserver* bandwidth_observer) override;
  void ResetSenderCongestionControlObjects() override;
  void SetRTCP_CNAME(absl::string_view c_name) override;
  std::vector<ReportBlock> 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;

  void OnUplinkPacketLossRate(float packet_loss_rate);

 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,
                       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_;
  uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);

  // uses
  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
  bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_);
  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
  // VoeRTP_RTCP
  // TODO(henrika): can today be accessed on the main thread and on the
  // task queue; hence potential race.
  bool _includeAudioLevelIndication;

  // RtcpBandwidthObserver
  const std::unique_ptr<VoERtcpObserver> rtcp_observer_;

  PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
      nullptr;
  TransportFeedbackObserver* const feedback_observer_;
  const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
  const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;

  SequenceChecker construction_thread_;

  bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;

  // 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_);

  const bool fixing_timestamp_stall_;

  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_);
    rtp_packet_pacer_->EnqueuePackets(std::move(packets));
  }

  void RemovePacketsForSsrc(uint32_t ssrc) override {
    MutexLock lock(&mutex_);
    rtp_packet_pacer_->RemovePacketsForSsrc(ssrc);
  }

 private:
  SequenceChecker thread_checker_;
  Mutex mutex_;
  RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&mutex_);
};

class VoERtcpObserver : public RtcpBandwidthObserver {
 public:
  explicit VoERtcpObserver(ChannelSend* owner)
      : owner_(owner), bandwidth_observer_(nullptr) {}
  ~VoERtcpObserver() override {}

  void SetBandwidthObserver(RtcpBandwidthObserver* bandwidth_observer) {
    MutexLock lock(&mutex_);
    bandwidth_observer_ = bandwidth_observer;
  }

  void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
    MutexLock lock(&mutex_);
    if (bandwidth_observer_) {
      bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
    }
  }

  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
                                    int64_t rtt,
                                    int64_t now_ms) override {
    {
      MutexLock lock(&mutex_);
      if (bandwidth_observer_) {
        bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, rtt,
                                                          now_ms);
      }
    }
    // TODO(mflodman): Do we need to aggregate reports here or can we jut send
    // what we get? I.e. do we ever get multiple reports bundled into one RTCP
    // report for VoiceEngine?
    if (report_blocks.empty())
      return;

    int fraction_lost_aggregate = 0;
    int total_number_of_packets = 0;

    // If receiving multiple report blocks, calculate the weighted average based
    // on the number of packets a report refers to.
    for (ReportBlockList::const_iterator block_it = report_blocks.begin();
         block_it != report_blocks.end(); ++block_it) {
      // Find the previous extended high sequence number for this remote SSRC,
      // to calculate the number of RTP packets this report refers to. Ignore if
      // we haven't seen this SSRC before.
      std::map<uint32_t, uint32_t>::iterator seq_num_it =
          extended_max_sequence_number_.find(block_it->source_ssrc);
      int number_of_packets = 0;
      if (seq_num_it != extended_max_sequence_number_.end()) {
        number_of_packets =
            block_it->extended_highest_sequence_number - seq_num_it->second;
      }
      fraction_lost_aggregate += number_of_packets * block_it->fraction_lost;
      total_number_of_packets += number_of_packets;

      extended_max_sequence_number_[block_it->source_ssrc] =
          block_it->extended_highest_sequence_number;
    }
    int weighted_fraction_lost = 0;
    if (total_number_of_packets > 0) {
      weighted_fraction_lost =
          (fraction_lost_aggregate + total_number_of_packets / 2) /
          total_number_of_packets;
    }
    owner_->OnUplinkPacketLossRate(weighted_fraction_lost / 255.0f);
  }

 private:
  ChannelSend* owner_;
  // Maps remote side ssrc to extended highest sequence number received.
  std::map<uint32_t, uint32_t> extended_max_sequence_number_;
  Mutex mutex_;
  RtcpBandwidthObserver* bandwidth_observer_ 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,
                                  rtc::ArrayView<const uint8_t> payload,
                                  int64_t absolute_capture_timestamp_ms) {
  if (_includeAudioLevelIndication) {
    // Store current audio level in the RTP sender.
    // The level will be used in combination with voice-activity state
    // (frameType) to add an RTP header extension
    rtp_sender_audio_->SetAudioLevel(rms_level_.Average());
  }

  // 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,
                                    // 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.
  if (!rtp_sender_audio_->SendAudio(
          frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
          payload.data(), payload.size(), absolute_capture_timestamp_ms)) {
    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,
    TransportFeedbackObserver* feedback_observer,
    const FieldTrialsView& field_trials)
    : ssrc_(ssrc),
      event_log_(rtc_event_log),
      _timeStamp(0),  // This is just an offset, RTP module will add it's own
                      // random offset
      input_mute_(false),
      previous_frame_muted_(false),
      _includeAudioLevelIndication(false),
      rtcp_observer_(new VoERtcpObserver(this)),
      feedback_observer_(feedback_observer),
      rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
      retransmission_rate_limiter_(
          new RateLimiter(clock, kMaxRetransmissionWindowMs)),
      frame_encryptor_(frame_encryptor),
      crypto_options_(crypto_options),
      fixing_timestamp_stall_(
          field_trials.IsDisabled("WebRTC-Audio-FixTimestampStall")),
      encoder_queue_(task_queue_factory->CreateTaskQueue(
          "AudioEncoder",
          TaskQueueFactory::Priority::NORMAL)) {
  audio_coding_.reset(AudioCodingModule::Create(AudioCodingModule::Config()));

  RtpRtcpInterface::Configuration configuration;
  configuration.bandwidth_callback = rtcp_observer_.get();
  configuration.transport_feedback_callback = 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;
  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);
}

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.
  encoder_queue_.PostTask([this] {
    RTC_DCHECK_RUN_ON(&encoder_queue_);
    encoder_queue_is_active_ = true;
  });
}

void ChannelSend::StopSend() {
  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
  if (!sending_) {
    return;
  }
  sending_ = false;

  rtc::Event flush;
  encoder_queue_.PostTask([this, &flush]() {
    RTC_DCHECK_RUN_ON(&encoder_queue_);
    encoder_queue_is_active_ = false;
    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::OnUplinkPacketLossRate(float packet_loss_rate) {
  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(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_);
  _includeAudioLevelIndication = enable;
  if (enable) {
    rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevel::Uri(), id);
  } else {
    rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevel::Uri());
  }
}

void ChannelSend::RegisterSenderCongestionControlObjects(
    RtpTransportControllerSendInterface* transport,
    RtcpBandwidthObserver* bandwidth_observer) {
  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_);
  rtcp_observer_->SetBandwidthObserver(bandwidth_observer);
  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);
  rtcp_observer_->SetBandwidthObserver(nullptr);
  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<ReportBlock> 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.
  std::vector<ReportBlock> report_blocks;
  for (const ReportBlockData& data : rtp_rtcp_->GetLatestReportBlockData()) {
    ReportBlock report_block;
    report_block.sender_SSRC = data.report_block().sender_ssrc;
    report_block.source_SSRC = data.report_block().source_ssrc;
    report_block.fraction_lost = data.report_block().fraction_lost;
    report_block.cumulative_num_packets_lost = data.report_block().packets_lost;
    report_block.extended_highest_sequence_number =
        data.report_block().extended_highest_sequence_number;
    report_block.interarrival_jitter = data.report_block().jitter;
    report_block.last_SR_timestamp =
        data.report_block().last_sender_report_timestamp;
    report_block.delay_since_last_SR =
        data.report_block().delay_since_last_sender_report;
    report_blocks.push_back(report_block);
  }
  return report_blocks;
}

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_rcvd = 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);

  // 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_) {
          if (fixing_timestamp_stall_) {
            _timeStamp +=
                static_cast<uint32_t>(audio_frame->samples_per_channel_);
          }
          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 (_includeAudioLevelIndication) {
          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;

        // Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.

        // The ACM resamples internally.
        audio_frame->timestamp_ = _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;
        }

        _timeStamp += static_cast<uint32_t>(audio_frame->samples_per_channel_);
      });
}

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, rtc::ArrayView<const uint8_t> payload,
             int64_t absolute_capture_timestamp_ms) {
        RTC_DCHECK_RUN_ON(&encoder_queue_);
        return SendRtpAudio(frameType, payloadType, rtp_timestamp, 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,
    TransportFeedbackObserver* feedback_observer,
    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),
      feedback_observer, field_trials);
}

}  // namespace voe
}  // namespace webrtc