blob: 7890d9b1e04067fe4f7ce6ffad8bc936066b65eb [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/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"
#include "rtc_base/logging.h"
#include "rtc_base/task_utils/to_queued_task.h"
namespace webrtc {
AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
std::vector<std::string> stream_ids,
bool is_unified_plan)
: AudioRtpReceiver(worker_thread,
receiver_id,
CreateStreamsFromIds(std::move(stream_ids)),
is_unified_plan) {}
AudioRtpReceiver::AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
bool is_unified_plan)
: 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_))),
cached_track_enabled_(track_->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(stopped_);
RTC_DCHECK(!media_channel_);
track_->GetSource()->UnregisterAudioObserver(this);
track_->UnregisterObserver(this);
}
void AudioRtpReceiver::OnChanged() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
worker_thread_->PostTask(ToQueuedTask(
worker_thread_safety_,
[this, enabled = cached_track_enabled_, volume = cached_volume_]() {
RTC_DCHECK_RUN_ON(worker_thread_);
Reconfigure(enabled, volume);
}));
}
}
// RTC_RUN_ON(worker_thread_)
void AudioRtpReceiver::SetOutputVolume_w(double volume) {
RTC_DCHECK_GE(volume, 0.0);
RTC_DCHECK_LE(volume, 10.0);
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);
// 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;
if (stopped_)
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 (track_->enabled()) {
worker_thread_->PostTask(
ToQueuedTask(worker_thread_safety_, [this, volume = cached_volume_]() {
RTC_DCHECK_RUN_ON(worker_thread_);
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_);
// TODO(deadbeef): Need to do more here to fully stop receiving packets.
if (!stopped_) {
source_->SetState(MediaSourceInterface::kEnded);
stopped_ = true;
}
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&]() {
RTC_DCHECK_RUN_ON(worker_thread_);
if (media_channel_)
SetOutputVolume_w(0.0);
SetMediaChannel_w(nullptr);
});
}
void AudioRtpReceiver::StopAndEndTrack() {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
Stop();
track_->internal()->set_ended();
}
void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) {
RTC_DCHECK_RUN_ON(&signaling_thread_checker_);
bool ok = worker_thread_->Invoke<bool>(
RTC_FROM_HERE, [&, enabled = cached_track_enabled_,
volume = cached_volume_, was_stopped = stopped_]() {
RTC_DCHECK_RUN_ON(worker_thread_);
if (!media_channel_) {
RTC_DCHECK(was_stopped);
return false; // Can't restart.
}
if (!was_stopped && ssrc_ == ssrc) {
// Already running with that ssrc.
RTC_DCHECK(worker_thread_safety_->alive());
return true;
}
if (!was_stopped) {
source_->Stop(media_channel_, ssrc_);
}
ssrc_ = std::move(ssrc);
source_->Start(media_channel_, ssrc_);
if (ssrc_) {
media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs());
}
Reconfigure(enabled, volume);
return true;
});
if (!ok)
return;
stopped_ = false;
}
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(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 {
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);
}
// RTC_RUN_ON(worker_thread_)
void AudioRtpReceiver::Reconfigure(bool track_enabled, double volume) {
RTC_DCHECK(media_channel_);
SetOutputVolume_w(track_enabled ? 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(&signaling_thread_checker_);
RTC_DCHECK(media_channel == nullptr ||
media_channel->media_type() == media_type());
if (stopped_ && !media_channel)
return;
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(worker_thread_);
SetMediaChannel_w(media_channel);
});
}
// RTC_RUN_ON(worker_thread_)
void AudioRtpReceiver::SetMediaChannel_w(cricket::MediaChannel* media_channel) {
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