292 lines
9.1 KiB
C++
292 lines
9.1 KiB
C++
/*
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* Copyright 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/audio_rtp_receiver.h"
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#include <stddef.h>
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#include <utility>
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#include <vector>
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#include "api/media_stream_proxy.h"
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#include "api/media_stream_track_proxy.h"
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#include "pc/audio_track.h"
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#include "pc/jitter_buffer_delay.h"
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#include "pc/jitter_buffer_delay_proxy.h"
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#include "pc/media_stream.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/location.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
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std::string receiver_id,
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std::vector<std::string> stream_ids)
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: AudioRtpReceiver(worker_thread,
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receiver_id,
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CreateStreamsFromIds(std::move(stream_ids))) {}
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AudioRtpReceiver::AudioRtpReceiver(
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rtc::Thread* worker_thread,
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const std::string& receiver_id,
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
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: worker_thread_(worker_thread),
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id_(receiver_id),
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source_(new rtc::RefCountedObject<RemoteAudioSource>(worker_thread)),
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track_(AudioTrackProxy::Create(rtc::Thread::Current(),
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AudioTrack::Create(receiver_id, source_))),
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cached_track_enabled_(track_->enabled()),
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attachment_id_(GenerateUniqueId()),
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delay_(JitterBufferDelayProxy::Create(
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rtc::Thread::Current(),
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worker_thread_,
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new rtc::RefCountedObject<JitterBufferDelay>(worker_thread))) {
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RTC_DCHECK(worker_thread_);
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RTC_DCHECK(track_->GetSource()->remote());
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track_->RegisterObserver(this);
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track_->GetSource()->RegisterAudioObserver(this);
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SetStreams(streams);
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}
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AudioRtpReceiver::~AudioRtpReceiver() {
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track_->GetSource()->UnregisterAudioObserver(this);
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track_->UnregisterObserver(this);
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Stop();
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}
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void AudioRtpReceiver::OnChanged() {
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if (cached_track_enabled_ != track_->enabled()) {
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cached_track_enabled_ = track_->enabled();
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Reconfigure();
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}
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}
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bool AudioRtpReceiver::SetOutputVolume(double volume) {
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RTC_DCHECK_GE(volume, 0.0);
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RTC_DCHECK_LE(volume, 10.0);
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RTC_DCHECK(media_channel_);
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RTC_DCHECK(!stopped_);
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return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
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return ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume)
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: media_channel_->SetDefaultOutputVolume(volume);
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});
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}
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void AudioRtpReceiver::OnSetVolume(double volume) {
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RTC_DCHECK_GE(volume, 0);
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RTC_DCHECK_LE(volume, 10);
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cached_volume_ = volume;
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if (!media_channel_ || stopped_) {
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RTC_LOG(LS_ERROR)
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<< "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
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return;
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}
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// When the track is disabled, the volume of the source, which is the
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// corresponding WebRtc Voice Engine channel will be 0. So we do not allow
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// setting the volume to the source when the track is disabled.
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if (!stopped_ && track_->enabled()) {
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if (!SetOutputVolume(cached_volume_)) {
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RTC_NOTREACHED();
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}
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}
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}
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std::vector<std::string> AudioRtpReceiver::stream_ids() const {
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std::vector<std::string> stream_ids(streams_.size());
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for (size_t i = 0; i < streams_.size(); ++i)
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stream_ids[i] = streams_[i]->id();
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return stream_ids;
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}
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RtpParameters AudioRtpReceiver::GetParameters() const {
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if (!media_channel_ || stopped_) {
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return RtpParameters();
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}
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return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
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return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_)
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: media_channel_->GetDefaultRtpReceiveParameters();
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});
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}
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void AudioRtpReceiver::SetFrameDecryptor(
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rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
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frame_decryptor_ = std::move(frame_decryptor);
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// Special Case: Set the frame decryptor to any value on any existing channel.
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if (media_channel_ && ssrc_.has_value() && !stopped_) {
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worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
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media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
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});
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}
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}
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rtc::scoped_refptr<FrameDecryptorInterface>
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AudioRtpReceiver::GetFrameDecryptor() const {
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return frame_decryptor_;
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}
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void AudioRtpReceiver::Stop() {
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// TODO(deadbeef): Need to do more here to fully stop receiving packets.
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if (stopped_) {
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return;
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}
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if (media_channel_) {
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// Allow that SetOutputVolume fail. This is the normal case when the
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// underlying media channel has already been deleted.
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SetOutputVolume(0.0);
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}
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stopped_ = true;
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}
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void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
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RTC_DCHECK(media_channel_);
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if (!stopped_ && ssrc_ == ssrc) {
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return;
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}
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if (!stopped_) {
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source_->Stop(media_channel_, ssrc_);
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delay_->OnStop();
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}
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ssrc_ = ssrc;
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stopped_ = false;
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source_->Start(media_channel_, ssrc);
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delay_->OnStart(media_channel_, ssrc.value_or(0));
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Reconfigure();
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}
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void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
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if (!media_channel_) {
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RTC_LOG(LS_ERROR)
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<< "AudioRtpReceiver::SetupMediaChannel: No audio channel exists.";
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return;
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}
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RestartMediaChannel(ssrc);
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}
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void AudioRtpReceiver::SetupUnsignaledMediaChannel() {
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if (!media_channel_) {
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RTC_LOG(LS_ERROR) << "AudioRtpReceiver::SetupUnsignaledMediaChannel: No "
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"audio channel exists.";
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}
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RestartMediaChannel(absl::nullopt);
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}
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void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
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SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
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}
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void AudioRtpReceiver::SetStreams(
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
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// Remove remote track from any streams that are going away.
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for (const auto& existing_stream : streams_) {
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bool removed = true;
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for (const auto& stream : streams) {
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if (existing_stream->id() == stream->id()) {
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RTC_DCHECK_EQ(existing_stream.get(), stream.get());
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removed = false;
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break;
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}
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}
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if (removed) {
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existing_stream->RemoveTrack(track_);
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}
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}
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// Add remote track to any streams that are new.
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for (const auto& stream : streams) {
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bool added = true;
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for (const auto& existing_stream : streams_) {
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if (stream->id() == existing_stream->id()) {
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RTC_DCHECK_EQ(stream.get(), existing_stream.get());
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added = false;
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break;
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}
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}
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if (added) {
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stream->AddTrack(track_);
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}
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}
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streams_ = streams;
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}
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std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
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if (!media_channel_ || !ssrc_ || stopped_) {
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return {};
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}
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return worker_thread_->Invoke<std::vector<RtpSource>>(
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RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
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}
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void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
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rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
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worker_thread_->Invoke<void>(
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RTC_FROM_HERE, [this, frame_transformer = std::move(frame_transformer)] {
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RTC_DCHECK_RUN_ON(worker_thread_);
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frame_transformer_ = frame_transformer;
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if (media_channel_ && ssrc_.has_value() && !stopped_) {
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media_channel_->SetDepacketizerToDecoderFrameTransformer(
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*ssrc_, frame_transformer);
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}
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});
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}
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void AudioRtpReceiver::Reconfigure() {
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if (!media_channel_ || stopped_) {
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RTC_LOG(LS_ERROR)
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<< "AudioRtpReceiver::Reconfigure: No audio channel exists.";
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return;
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}
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if (!SetOutputVolume(track_->enabled() ? cached_volume_ : 0)) {
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RTC_NOTREACHED();
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}
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// Reattach the frame decryptor if we were reconfigured.
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MaybeAttachFrameDecryptorToMediaChannel(
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ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
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if (media_channel_ && ssrc_.has_value() && !stopped_) {
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worker_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
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RTC_DCHECK_RUN_ON(worker_thread_);
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if (!frame_transformer_)
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return;
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media_channel_->SetDepacketizerToDecoderFrameTransformer(
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*ssrc_, frame_transformer_);
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});
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}
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}
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void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
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observer_ = observer;
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// Deliver any notifications the observer may have missed by being set late.
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if (received_first_packet_ && observer_) {
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observer_->OnFirstPacketReceived(media_type());
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}
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}
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void AudioRtpReceiver::SetJitterBufferMinimumDelay(
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absl::optional<double> delay_seconds) {
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delay_->Set(delay_seconds);
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}
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void AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
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RTC_DCHECK(media_channel == nullptr ||
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media_channel->media_type() == media_type());
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media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
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}
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void AudioRtpReceiver::NotifyFirstPacketReceived() {
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if (observer_) {
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observer_->OnFirstPacketReceived(media_type());
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}
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received_first_packet_ = true;
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}
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} // namespace webrtc
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