blob: da2f81b162b0cc623d0caa4a0e581011d5c2ab9c [file] [log] [blame]
/*
* Copyright 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 "pc/rtpreceiver.h"
#include <utility>
#include <vector>
#include "api/mediastreamproxy.h"
#include "api/mediastreamtrackproxy.h"
#include "api/videosourceproxy.h"
#include "pc/audiotrack.h"
#include "pc/mediastream.h"
#include "pc/videotrack.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
// This function is only expected to be called on the signalling thread.
int GenerateUniqueId() {
static int g_unique_id = 0;
return ++g_unique_id;
}
std::vector<rtc::scoped_refptr<MediaStreamInterface>> CreateStreamsFromIds(
std::vector<std::string> stream_ids) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams(
stream_ids.size());
for (size_t i = 0; i < stream_ids.size(); ++i) {
streams[i] = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(std::move(stream_ids[i])));
}
return streams;
}
} // namespace
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_(AudioTrackProxy::Create(rtc::Thread::Current(),
AudioTrack::Create(receiver_id, source_))),
cached_track_enabled_(track_->enabled()),
attachment_id_(GenerateUniqueId()) {
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(ssrc_);
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetOutputVolume(*ssrc_, volume);
});
}
void AudioRtpReceiver::OnSetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
cached_volume_ = volume;
if (!media_channel_ || !ssrc_) {
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_ || !ssrc_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return media_channel_->GetRtpReceiveParameters(*ssrc_);
});
}
bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
if (!media_channel_ || !ssrc_ || stopped_) {
return false;
}
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetRtpReceiveParameters(*ssrc_, parameters);
});
}
void AudioRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
if (media_channel_ && ssrc_) {
// 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::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::SetupMediaChannel: No audio channel exists.";
return;
}
if (ssrc_ == ssrc) {
return;
}
if (ssrc_) {
source_->Stop(media_channel_, *ssrc_);
}
ssrc_ = ssrc;
source_->Start(media_channel_, *ssrc_);
Reconfigure();
}
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 (auto existing_stream : streams_) {
bool removed = true;
for (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 (auto stream : streams) {
bool added = true;
for (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::Reconfigure() {
RTC_DCHECK(!stopped_);
if (!media_channel_ || !ssrc_) {
RTC_LOG(LS_ERROR)
<< "AudioRtpReceiver::Reconfigure: No audio channel exists.";
return;
}
if (!SetOutputVolume(track_->enabled() ? cached_volume_ : 0)) {
RTC_NOTREACHED();
}
}
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::NotifyFirstPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
VideoRtpReceiver::VideoRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids)
: VideoRtpReceiver(worker_thread,
receiver_id,
CreateStreamsFromIds(std::move(stream_ids))) {}
VideoRtpReceiver::VideoRtpReceiver(
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 RefCountedObject<VideoRtpTrackSource>()),
track_(VideoTrackProxy::Create(
rtc::Thread::Current(),
worker_thread,
VideoTrack::Create(
receiver_id,
VideoTrackSourceProxy::Create(rtc::Thread::Current(),
worker_thread,
source_),
worker_thread))),
attachment_id_(GenerateUniqueId()) {
RTC_DCHECK(worker_thread_);
SetStreams(streams);
source_->SetState(MediaSourceInterface::kLive);
}
VideoRtpReceiver::~VideoRtpReceiver() {
// Since cricket::VideoRenderer is not reference counted,
// we need to remove it from the channel before we are deleted.
Stop();
}
std::vector<std::string> VideoRtpReceiver::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;
}
bool VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
RTC_DCHECK(media_channel_);
RTC_DCHECK(ssrc_);
return worker_thread_->Invoke<bool>(
RTC_FROM_HERE, [&] { return media_channel_->SetSink(*ssrc_, sink); });
}
RtpParameters VideoRtpReceiver::GetParameters() const {
if (!media_channel_ || !ssrc_ || stopped_) {
return RtpParameters();
}
return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
return media_channel_->GetRtpReceiveParameters(*ssrc_);
});
}
bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
if (!media_channel_ || !ssrc_ || stopped_) {
return false;
}
return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
return media_channel_->SetRtpReceiveParameters(*ssrc_, parameters);
});
}
void VideoRtpReceiver::Stop() {
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (stopped_) {
return;
}
source_->SetState(MediaSourceInterface::kEnded);
if (!media_channel_ || !ssrc_) {
RTC_LOG(LS_WARNING) << "VideoRtpReceiver::Stop: No video channel exists.";
} else {
// Allow that SetSink fail. This is the normal case when the underlying
// media channel has already been deleted.
SetSink(nullptr);
}
stopped_ = true;
}
void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
if (!media_channel_) {
RTC_LOG(LS_ERROR)
<< "VideoRtpReceiver::SetupMediaChannel: No video channel exists.";
}
if (ssrc_ == ssrc) {
return;
}
if (ssrc_) {
SetSink(nullptr);
}
ssrc_ = ssrc;
SetSink(source_->sink());
}
void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
}
void VideoRtpReceiver::SetStreams(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
// Remove remote track from any streams that are going away.
for (auto existing_stream : streams_) {
bool removed = true;
for (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 (auto stream : streams) {
bool added = true;
for (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;
}
void VideoRtpReceiver::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 VideoRtpReceiver::NotifyFirstPacketReceived() {
if (observer_) {
observer_->OnFirstPacketReceived(media_type());
}
received_first_packet_ = true;
}
} // namespace webrtc