1 files changed, 514 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/audio_receive_stream.cc b/third_party/libwebrtc/audio/audio_receive_stream.cc
new file mode 100644
index 0000000000..93e2dd99d1
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_receive_stream.cc
@@ -0,0 +1,514 @@
+/*
+ *  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 << ", transport_cc: " << (transport_cc ? "on" : "off");
+  ss << ", nack: " << nack.ToString();
+  ss << ", extensions: [";
+  for (size_t i = 0; i < extensions.size(); ++i) {
+    ss << extensions[i].ToString();
+    if (i != extensions.size() - 1) {
+      ss << ", ";
+    }
+  }
+  ss << ']';
+  ss << ", rtcp_event_observer: "
+     << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr");
+  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),
+      config.rtp.rtcp_event_observer);
+}
+}  // 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_);
+
+  packet_sequence_checker_.Detach();
+
+  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;
+  }
+  channel_receive_->StartPlayout();
+  playing_ = true;
+  audio_state()->AddReceivingStream(this);
+}
+
+void AudioReceiveStreamImpl::Stop() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!playing_) {
+    return;
+  }
+  channel_receive_->StopPlayout();
+  playing_ = false;
+  audio_state()->RemoveReceivingStream(this);
+}
+
+bool AudioReceiveStreamImpl::transport_cc() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return config_.rtp.transport_cc;
+}
+
+void AudioReceiveStreamImpl::SetTransportCc(bool transport_cc) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  config_.rtp.transport_cc = transport_cc;
+}
+
+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));
+}
+
+void AudioReceiveStreamImpl::SetRtpExtensions(
+    std::vector<RtpExtension> extensions) {
+  // 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_);
+  config_.rtp.extensions = std::move(extensions);
+}
+
+const std::vector<RtpExtension>& AudioReceiveStreamImpl::GetRtpExtensions()
+    const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_.rtp.extensions;
+}
+
+RtpHeaderExtensionMap AudioReceiveStreamImpl::GetRtpExtensionMap() const {
+  return RtpHeaderExtensionMap(config_.rtp.extensions);
+}
+
+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_rcvd = call_stats.payload_bytes_rcvd;
+  stats.header_and_padding_bytes_rcvd =
+      call_stats.header_and_padding_bytes_rcvd;
+  stats.packets_rcvd = 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_timestamp_ms =
+      call_stats.last_packet_received_timestamp_ms;
+  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 {
+  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) {
+    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