blob: 8ff685d8e2d4a7e7c24d5f4471870e314b826656 [file] [log] [blame]
/*
* 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 <utility>
#include <vector>
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "pc/audio_track.h"
#include "pc/jitter_buffer_delay.h"
#include "pc/jitter_buffer_delay_proxy.h"
#include "pc/media_stream.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids)
: AudioRtpReceiver(worker_thread,
receiver_id,
CreateStreamsFromIds(std::move(stream_ids))) {}
AudioRtpReceiver::AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
: worker_thread_(worker_thread),
id_(receiver_id),
source_(new rtc::RefCountedObject<RemoteAudioSource>(worker_thread)),
track_(AudioTrackProxyWithInternal<AudioTrack>::Create(
rtc::Thread::Current(),
AudioTrack::Create(receiver_id, source_))),
cached_track_enabled_(track_->enabled()),
attachment_id_(GenerateUniqueId()),
delay_(JitterBufferDelayProxy::Create(
rtc::Thread::Current(),
worker_thread_,
new rtc::RefCountedObject<JitterBufferDelay>(worker_thread))) {
RTC_DCHECK(worker_thread_);
RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
SetStreams(streams);
}
AudioRtpReceiver::~AudioRtpReceiver() {
track_->GetSource()->UnregisterAudioObserver(this);
track_->UnregisterObserver(this);
Stop();
}
void AudioRtpReceiver::OnChanged() {
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
Reconfigure();
}
}
bool AudioRtpReceiver::SetOutputVolume(double volume) {
RTC_DCHECK_GE(volume, 0.0);
RTC_DCHECK_LE(volume, 10.0);
RTC_DCHECK(media_channel_);
RTC_DCHECK(!stopped_);
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume)
: media_channel_->SetDefaultOutputVolume(volume);
});
}
void AudioRtpReceiver::OnSetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
cached_volume_ = volume;
if (!media_channel_ || stopped_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
return;
}
// 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 (!stopped_ && track_->enabled()) {
if (!SetOutputVolume(cached_volume_)) {
RTC_NOTREACHED();
}
}
}
std::vector<std::string> AudioRtpReceiver::stream_ids() const {
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;
}
RtpParameters AudioRtpReceiver::GetParameters() const {
if (!media_channel_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_)
: media_channel_->GetDefaultRtpReceiveParameters();
});
}
void AudioRtpReceiver::SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
frame_decryptor_ = std::move(frame_decryptor);
// Special Case: Set the frame decryptor to any value on any existing channel.
if (media_channel_ && ssrc_.has_value() && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_);
});
}
}
rtc::scoped_refptr<FrameDecryptorInterface>
AudioRtpReceiver::GetFrameDecryptor() const {
return frame_decryptor_;
}
void AudioRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
if (media_channel_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
SetOutputVolume(0.0);
}
stopped_ = true;
}
void AudioRtpReceiver::StopAndEndTrack() {
Stop();
track_->internal()->set_ended();
}
void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
RTC_DCHECK(media_channel_);
if (!stopped_ && ssrc_ == ssrc) {
return;
}
if (!stopped_) {
source_->Stop(media_channel_, ssrc_);
delay_->OnStop();
}
ssrc_ = ssrc;
stopped_ = false;
source_->Start(media_channel_, ssrc);
delay_->OnStart(media_channel_, ssrc.value_or(0));
Reconfigure();
}
void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::SetupMediaChannel: No audio channel exists.";
return;
}
RestartMediaChannel(ssrc);
}
void AudioRtpReceiver::SetupUnsignaledMediaChannel() {
if (!media_channel_) {
RTC_LOG(LS_ERROR) << "AudioRtpReceiver::SetupUnsignaledMediaChannel: No "
"audio channel exists.";
}
RestartMediaChannel(absl::nullopt);
}
void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
}
void AudioRtpReceiver::SetStreams(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
// 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(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(track_);
}
}
streams_ = streams;
}
std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return {};
}
return worker_thread_->Invoke<std::vector<RtpSource>>(
RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
}
void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
worker_thread_->Invoke<void>(
RTC_FROM_HERE, [this, frame_transformer = std::move(frame_transformer)] {
RTC_DCHECK_RUN_ON(worker_thread_);
frame_transformer_ = frame_transformer;
if (media_channel_ && ssrc_.has_value() && !stopped_) {
media_channel_->SetDepacketizerToDecoderFrameTransformer(
*ssrc_, frame_transformer);
}
});
}
void AudioRtpReceiver::Reconfigure() {
if (!media_channel_ || stopped_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::Reconfigure: No audio channel exists.";
return;
}
if (!SetOutputVolume(track_->enabled() ? cached_volume_ : 0)) {
RTC_NOTREACHED();
}
// Reattach the frame decryptor if we were reconfigured.
MaybeAttachFrameDecryptorToMediaChannel(
ssrc_, worker_thread_, frame_decryptor_, media_channel_, stopped_);
if (media_channel_ && ssrc_.has_value() && !stopped_) {
worker_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!frame_transformer_)
return;
media_channel_->SetDepacketizerToDecoderFrameTransformer(
*ssrc_, frame_transformer_);
});
}
}
void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
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) {
delay_->Set(delay_seconds);
}
void AudioRtpReceiver::SetMediaChannel(cricket::MediaChannel* media_channel) {
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
media_channel_ = static_cast<cricket::VoiceMediaChannel*>(media_channel);
}
void AudioRtpReceiver::NotifyFirstPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
} // namespace webrtc