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// Copyright 2015 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/platform/webrtc/peer_connection_remote_audio_source.h"
#include <string>
#include <utility>
#include "base/check_op.h"
#include "base/strings/stringprintf.h"
#include "base/time/time.h"
#include "media/base/audio_bus.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
namespace blink {
namespace {
// Used as an identifier for the down-casters.
void* const kPeerConnectionRemoteTrackIdentifier =
const_cast<void**>(&kPeerConnectionRemoteTrackIdentifier);
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("PCRAS::" + message);
}
} // namespace
PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack(
scoped_refptr<webrtc::AudioTrackInterface> track_interface)
: MediaStreamAudioTrack(false /* is_local_track */),
track_interface_(std::move(track_interface)) {
blink::WebRtcLogMessage(
base::StringPrintf("PCRAT::PeerConnectionRemoteAudioTrack({id=%s})",
track_interface_->id().c_str()));
}
PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack() {
blink::WebRtcLogMessage(
base::StringPrintf("PCRAT::~PeerConnectionRemoteAudioTrack([id=%s])",
track_interface_->id().c_str()));
// Ensure the track is stopped.
MediaStreamAudioTrack::Stop();
}
// static
PeerConnectionRemoteAudioTrack* PeerConnectionRemoteAudioTrack::From(
MediaStreamAudioTrack* track) {
if (track &&
track->GetClassIdentifier() == kPeerConnectionRemoteTrackIdentifier)
return static_cast<PeerConnectionRemoteAudioTrack*>(track);
return nullptr;
}
void PeerConnectionRemoteAudioTrack::SetEnabled(bool enabled) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
blink::WebRtcLogMessage(base::StringPrintf(
"PCRAT::SetEnabled([id=%s] {enabled=%s})", track_interface_->id().c_str(),
(enabled ? "true" : "false")));
// This affects the shared state of the source for whether or not it's a part
// of the mixed audio that's rendered for remote tracks from WebRTC.
// All tracks from the same source will share this state and thus can step
// on each other's toes.
// This is also why we can't check the enabled state for equality with
// |enabled| before setting the mixing enabled state. This track's enabled
// state and the shared state might not be the same.
track_interface_->set_enabled(enabled);
MediaStreamAudioTrack::SetEnabled(enabled);
}
void* PeerConnectionRemoteAudioTrack::GetClassIdentifier() const {
return kPeerConnectionRemoteTrackIdentifier;
}
PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource(
scoped_refptr<webrtc::AudioTrackInterface> track_interface,
scoped_refptr<base::SingleThreadTaskRunner> task_runner)
: MediaStreamAudioSource(std::move(task_runner),
false /* is_local_source */),
track_interface_(std::move(track_interface)),
is_sink_of_peer_connection_(false) {
DCHECK(track_interface_);
SendLogMessage(base::StringPrintf("PeerConnectionRemoteAudioSource([id=%s])",
track_interface_->id().c_str()));
}
PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource() {
SendLogMessage(base::StringPrintf("~PeerConnectionRemoteAudioSource([id=%s])",
track_interface_->id().c_str()));
EnsureSourceIsStopped();
}
std::unique_ptr<MediaStreamAudioTrack>
PeerConnectionRemoteAudioSource::CreateMediaStreamAudioTrack(
const std::string& id) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
return std::make_unique<PeerConnectionRemoteAudioTrack>(track_interface_);
}
bool PeerConnectionRemoteAudioSource::EnsureSourceIsStarted() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (is_sink_of_peer_connection_)
return true;
SendLogMessage(base::StringPrintf("EnsureSourceIsStarted([id=%s])",
track_interface_->id().c_str()));
track_interface_->AddSink(this);
is_sink_of_peer_connection_ = true;
return true;
}
void PeerConnectionRemoteAudioSource::EnsureSourceIsStopped() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (is_sink_of_peer_connection_) {
SendLogMessage(base::StringPrintf("EnsureSourceIsStopped([id=%s])",
track_interface_->id().c_str()));
track_interface_->RemoveSink(this);
is_sink_of_peer_connection_ = false;
}
}
void PeerConnectionRemoteAudioSource::OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {
// Debug builds: Note that this lock isn't meant to synchronize anything.
// Instead, it is being used as a run-time check to ensure there isn't already
// another thread executing this method. The reason we don't use
// base::ThreadChecker here is because we shouldn't be making assumptions
// about the private threading model of libjingle. For example, it would be
// legitimate for libjingle to use a different thread to invoke this method
// whenever the audio format changes.
#ifndef NDEBUG
const bool is_only_thread_here = single_audio_thread_guard_.Try();
DCHECK(is_only_thread_here);
#endif
// TODO(tommi): We should get the timestamp from WebRTC.
base::TimeTicks playout_time(base::TimeTicks::Now());
if (!audio_bus_ ||
static_cast<size_t>(audio_bus_->channels()) != number_of_channels ||
static_cast<size_t>(audio_bus_->frames()) != number_of_frames) {
audio_bus_ = media::AudioBus::Create(number_of_channels, number_of_frames);
}
// Only 16 bits per sample is ever used. The FromInterleaved() call should
// be updated if that is no longer the case.
DCHECK_EQ(bits_per_sample, 16);
audio_bus_->FromInterleaved<media::SignedInt16SampleTypeTraits>(
reinterpret_cast<const int16_t*>(audio_data), number_of_frames);
media::AudioParameters params = MediaStreamAudioSource::GetAudioParameters();
if (!params.IsValid() ||
params.format() != media::AudioParameters::AUDIO_PCM_LOW_LATENCY ||
static_cast<size_t>(params.channels()) != number_of_channels ||
params.sample_rate() != sample_rate ||
static_cast<size_t>(params.frames_per_buffer()) != number_of_frames) {
MediaStreamAudioSource::SetFormat(
media::AudioParameters(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
media::GuessChannelLayout(number_of_channels),
sample_rate, number_of_frames));
}
MediaStreamAudioSource::DeliverDataToTracks(*audio_bus_, playout_time);
#ifndef NDEBUG
if (is_only_thread_here)
single_audio_thread_guard_.Release();
#endif
}
} // namespace blink
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