1 files changed, 338 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/audio_rtp_receiver.cc b/third_party/libwebrtc/pc/audio_rtp_receiver.cc
new file mode 100644
index 0000000000..e58b74ec72
--- /dev/null
+++ b/third_party/libwebrtc/pc/audio_rtp_receiver.cc
@@ -0,0 +1,338 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "pc/audio_rtp_receiver.h"
+
+#include <stddef.h>
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "pc/audio_track.h"
+#include "pc/media_stream_track_proxy.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/location.h"
+
+namespace webrtc {
+
+AudioRtpReceiver::AudioRtpReceiver(
+    rtc::Thread* worker_thread,
+    std::string receiver_id,
+    std::vector<std::string> stream_ids,
+    bool is_unified_plan,
+    cricket::VoiceMediaChannel* voice_channel /*= nullptr*/)
+    : AudioRtpReceiver(worker_thread,
+                       receiver_id,
+                       CreateStreamsFromIds(std::move(stream_ids)),
+                       is_unified_plan,
+                       voice_channel) {}
+
+AudioRtpReceiver::AudioRtpReceiver(
+    rtc::Thread* worker_thread,
+    const std::string& receiver_id,
+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
+    bool is_unified_plan,
+    cricket::VoiceMediaChannel* voice_channel /*= nullptr*/)
+    : worker_thread_(worker_thread),
+      id_(receiver_id),
+      source_(rtc::make_ref_counted<RemoteAudioSource>(
+          worker_thread,
+          is_unified_plan
+              ? RemoteAudioSource::OnAudioChannelGoneAction::kSurvive
+              : RemoteAudioSource::OnAudioChannelGoneAction::kEnd)),
+      track_(AudioTrackProxyWithInternal<AudioTrack>::Create(
+          rtc::Thread::Current(),
+          AudioTrack::Create(receiver_id, source_))),
+      media_channel_(voice_channel),
+      cached_track_enabled_(track_->internal()->enabled()),
+      attachment_id_(GenerateUniqueId()),
+      worker_thread_safety_(PendingTaskSafetyFlag::CreateDetachedInactive()) {
+  RTC_DCHECK(worker_thread_);
+  RTC_DCHECK(track_->GetSource()->remote());
+  track_->RegisterObserver(this);
+  track_->GetSource()->RegisterAudioObserver(this);
+  SetStreams(streams);
+}
+
+AudioRtpReceiver::~AudioRtpReceiver() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  RTC_DCHECK(!media_channel_);
+
+  track_->GetSource()->UnregisterAudioObserver(this);
+  track_->UnregisterObserver(this);
+}
+
+void AudioRtpReceiver::OnChanged() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  const bool enabled = track_->internal()->enabled();
+  if (cached_track_enabled_ == enabled)
+    return;
+  cached_track_enabled_ = enabled;
+  worker_thread_->PostTask(SafeTask(worker_thread_safety_, [this, enabled]() {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    Reconfigure(enabled);
+  }));
+}
+
+void AudioRtpReceiver::SetOutputVolume_w(double volume) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK_GE(volume, 0.0);
+  RTC_DCHECK_LE(volume, 10.0);
+
+  if (!media_channel_)
+    return;
+
+  ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume)
+        : media_channel_->SetDefaultOutputVolume(volume);
+}
+
+void AudioRtpReceiver::OnSetVolume(double volume) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  RTC_DCHECK_GE(volume, 0);
+  RTC_DCHECK_LE(volume, 10);
+
+  bool track_enabled = track_->internal()->enabled();
+  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    // Update the cached_volume_ even when stopped, to allow clients to set
+    // the volume before starting/restarting, eg see crbug.com/1272566.
+    cached_volume_ = volume;
+    // When the track is disabled, the volume of the source, which is the
+    // corresponding WebRtc Voice Engine channel will be 0. So we do not
+    // allow setting the volume to the source when the track is disabled.
+    if (track_enabled)
+      SetOutputVolume_w(volume);
+  });
+}
+
+rtc::scoped_refptr<DtlsTransportInterface> AudioRtpReceiver::dtls_transport()
+    const {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  return dtls_transport_;
+}
+
+std::vector<std::string> AudioRtpReceiver::stream_ids() const {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  std::vector<std::string> stream_ids(streams_.size());
+  for (size_t i = 0; i < streams_.size(); ++i)
+    stream_ids[i] = streams_[i]->id();
+  return stream_ids;
+}
+
+std::vector<rtc::scoped_refptr<MediaStreamInterface>>
+AudioRtpReceiver::streams() const {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  return streams_;
+}
+
+RtpParameters AudioRtpReceiver::GetParameters() const {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (!media_channel_)
+    return RtpParameters();
+  return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_)
+               : media_channel_->GetDefaultRtpReceiveParameters();
+}
+
+void AudioRtpReceiver::SetFrameDecryptor(
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  frame_decryptor_ = std::move(frame_decryptor);
+  // Special Case: Set the frame decryptor to any value on any existing channel.
+  if (media_channel_ && ssrc_) {
+    media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
+  }
+}
+
+rtc::scoped_refptr<FrameDecryptorInterface>
+AudioRtpReceiver::GetFrameDecryptor() const {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  return frame_decryptor_;
+}
+
+void AudioRtpReceiver::Stop() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  source_->SetState(MediaSourceInterface::kEnded);
+  track_->internal()->set_ended();
+}
+
+void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  bool enabled = track_->internal()->enabled();
+  MediaSourceInterface::SourceState state = source_->state();
+  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    RestartMediaChannel_w(std::move(ssrc), enabled, state);
+  });
+  source_->SetState(MediaSourceInterface::kLive);
+}
+
+void AudioRtpReceiver::RestartMediaChannel_w(
+    absl::optional<uint32_t> ssrc,
+    bool track_enabled,
+    MediaSourceInterface::SourceState state) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (!media_channel_)
+    return;  // Can't restart.
+
+  // Make sure the safety flag is marked as `alive` for cases where the media
+  // channel was provided via the ctor and not an explicit call to
+  // SetMediaChannel.
+  worker_thread_safety_->SetAlive();
+
+  if (state != MediaSourceInterface::kInitializing) {
+    if (ssrc_ == ssrc)
+      return;
+    source_->Stop(media_channel_, ssrc_);
+  }
+
+  ssrc_ = std::move(ssrc);
+  source_->Start(media_channel_, ssrc_);
+  if (ssrc_) {
+    media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
+  }
+
+  Reconfigure(track_enabled);
+}
+
+void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  RestartMediaChannel(ssrc);
+}
+
+void AudioRtpReceiver::SetupUnsignaledMediaChannel() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  RestartMediaChannel(absl::nullopt);
+}
+
+uint32_t AudioRtpReceiver::ssrc() const {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  return ssrc_.value_or(0);
+}
+
+void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
+}
+
+void AudioRtpReceiver::set_transport(
+    rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  dtls_transport_ = std::move(dtls_transport);
+}
+
+void AudioRtpReceiver::SetStreams(
+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  // Remove remote track from any streams that are going away.
+  for (const auto& existing_stream : streams_) {
+    bool removed = true;
+    for (const auto& stream : streams) {
+      if (existing_stream->id() == stream->id()) {
+        RTC_DCHECK_EQ(existing_stream.get(), stream.get());
+        removed = false;
+        break;
+      }
+    }
+    if (removed) {
+      existing_stream->RemoveTrack(audio_track());
+    }
+  }
+  // Add remote track to any streams that are new.
+  for (const auto& stream : streams) {
+    bool added = true;
+    for (const auto& existing_stream : streams_) {
+      if (stream->id() == existing_stream->id()) {
+        RTC_DCHECK_EQ(stream.get(), existing_stream.get());
+        added = false;
+        break;
+      }
+    }
+    if (added) {
+      stream->AddTrack(audio_track());
+    }
+  }
+  streams_ = streams;
+}
+
+std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (!media_channel_ || !ssrc_) {
+    return {};
+  }
+  return media_channel_->GetSources(*ssrc_);
+}
+
+void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (media_channel_) {
+    media_channel_->SetDepacketizerToDecoderFrameTransformer(ssrc_.value_or(0),
+                                                             frame_transformer);
+  }
+  frame_transformer_ = std::move(frame_transformer);
+}
+
+void AudioRtpReceiver::Reconfigure(bool track_enabled) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(media_channel_);
+
+  SetOutputVolume_w(track_enabled ? cached_volume_ : 0);
+
+  if (ssrc_ && frame_decryptor_) {
+    // Reattach the frame decryptor if we were reconfigured.
+    media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
+  }
+
+  if (frame_transformer_) {
+    media_channel_->SetDepacketizerToDecoderFrameTransformer(
+        ssrc_.value_or(0), frame_transformer_);
+  }
+}
+
+void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  observer_ = observer;
+  // Deliver any notifications the observer may have missed by being set late.
+  if (received_first_packet_ && observer_) {
+    observer_->OnFirstPacketReceived(media_type());
+  }
+}
+
+void AudioRtpReceiver::SetJitterBufferMinimumDelay(
+    absl::optional<double> delay_seconds) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  delay_.Set(delay_seconds);
+  if (media_channel_ && ssrc_)
+    media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
+}
+
+void AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(media_channel == nullptr ||
+             media_channel->media_type() == media_type());
+  if (!media_channel && media_channel_)
+    SetOutputVolume_w(0.0);
+
+  media_channel ? worker_thread_safety_->SetAlive()
+                : worker_thread_safety_->SetNotAlive();
+  media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
+}
+
+void AudioRtpReceiver::NotifyFirstPacketReceived() {
+  RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
+  if (observer_) {
+    observer_->OnFirstPacketReceived(media_type());
+  }
+  received_first_packet_ = true;
+}
+
+}  // namespace webrtc