blob: 63944c6d83ae219868ce1c99ac0061581c120350 [file] [log] [blame]
/*
* Copyright 2014 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/remote_audio_source.h"
#include <stddef.h>
#include <string>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "api/scoped_refptr.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
namespace {
constexpr int kDefaultLatency = 0;
constexpr int kRoundToZeroThresholdMs = 10;
} // namespace
// This proxy is passed to the underlying media engine to receive audio data as
// they come in. The data will then be passed back up to the RemoteAudioSource
// which will fan it out to all the sinks that have been added to it.
class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
public:
explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) {
RTC_DCHECK(source);
}
~AudioDataProxy() override { source_->OnAudioChannelGone(); }
// AudioSinkInterface implementation.
void OnData(const AudioSinkInterface::Data& audio) override {
source_->OnData(audio);
}
private:
const rtc::scoped_refptr<RemoteAudioSource> source_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioDataProxy);
};
RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
: main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
}
RemoteAudioSource::~RemoteAudioSource() {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(audio_observers_.empty());
RTC_DCHECK(sinks_.empty());
}
void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel,
uint32_t ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
// Check that there are no consecutive start calls.
RTC_DCHECK(!media_channel_ && !ssrc_);
// Remember media channel ssrc pair for latency calls.
media_channel_ = media_channel;
ssrc_ = ssrc;
// Register for callbacks immediately before AddSink so that we always get
// notified when a channel goes out of scope (signaled when "AudioDataProxy"
// is destroyed).
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel->SetRawAudioSink(ssrc,
absl::make_unique<AudioDataProxy>(this));
});
// Trying to apply cached latency for the audio stream.
if (cached_latency_) {
SetLatency(cached_latency_.value());
}
}
void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
uint32_t ssrc) {
RTC_DCHECK_RUN_ON(main_thread_);
RTC_DCHECK(media_channel);
// Assume that audio stream is no longer present for latency calls.
media_channel_ = nullptr;
ssrc_ = absl::nullopt;
worker_thread_->Invoke<void>(
RTC_FROM_HERE, [&] { media_channel->SetRawAudioSink(ssrc, nullptr); });
}
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK(main_thread_->IsCurrent());
return state_;
}
bool RemoteAudioSource::remote() const {
RTC_DCHECK(main_thread_->IsCurrent());
return true;
}
void RemoteAudioSource::SetVolume(double volume) {
RTC_DCHECK_GE(volume, 0);
RTC_DCHECK_LE(volume, 10);
for (auto* observer : audio_observers_) {
observer->OnSetVolume(volume);
}
}
void RemoteAudioSource::SetLatency(double latency) {
RTC_DCHECK_GE(latency, 0);
RTC_DCHECK_LE(latency, 10);
int delay_ms = rtc::dchecked_cast<int>(latency * 1000);
// In NetEq 0 delay has special meaning of being unconstrained value that is
// why we round delay to 0 if it is small enough during conversion from
// latency.
if (delay_ms <= kRoundToZeroThresholdMs) {
delay_ms = 0;
}
cached_latency_ = latency;
SetDelayMs(delay_ms);
}
double RemoteAudioSource::GetLatency() const {
absl::optional<int> delay_ms = GetDelayMs();
if (delay_ms) {
return delay_ms.value() / 1000.0;
} else {
return cached_latency_.value_or(kDefaultLatency);
}
}
bool RemoteAudioSource::SetDelayMs(int delay_ms) {
if (!media_channel_ || !ssrc_) {
return false;
}
worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms);
});
return true;
}
absl::optional<int> RemoteAudioSource::GetDelayMs() const {
if (!media_channel_ || !ssrc_) {
return absl::nullopt;
}
return worker_thread_->Invoke<absl::optional<int>>(RTC_FROM_HERE, [&] {
return media_channel_->GetBaseMinimumPlayoutDelayMs(ssrc_.value());
});
}
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
RTC_DCHECK(!absl::c_linear_search(audio_observers_, observer));
audio_observers_.push_back(observer);
}
void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
RTC_DCHECK(observer != NULL);
audio_observers_.remove(observer);
}
void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
if (state_ != MediaSourceInterface::kLive) {
RTC_LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
return;
}
rtc::CritScope lock(&sink_lock_);
RTC_DCHECK(!absl::c_linear_search(sinks_, sink));
sinks_.push_back(sink);
}
void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
RTC_DCHECK(main_thread_->IsCurrent());
RTC_DCHECK(sink);
rtc::CritScope lock(&sink_lock_);
sinks_.remove(sink);
}
void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
// Called on the externally-owned audio callback thread, via/from webrtc.
rtc::CritScope lock(&sink_lock_);
for (auto* sink : sinks_) {
sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
audio.samples_per_channel);
}
}
void RemoteAudioSource::OnAudioChannelGone() {
// Called when the audio channel is deleted. It may be the worker thread
// in libjingle or may be a different worker thread.
// This object needs to live long enough for the cleanup logic in OnMessage to
// run, so take a reference to it as the data. Sometimes the message may not
// be processed (because the thread was destroyed shortly after this call),
// but that is fine because the thread destructor will take care of destroying
// the message data which will release the reference on RemoteAudioSource.
main_thread_->Post(RTC_FROM_HERE, this, 0,
new rtc::ScopedRefMessageData<RemoteAudioSource>(this));
}
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
sinks_.clear();
state_ = MediaSourceInterface::kEnded;
FireOnChanged();
// Will possibly delete this RemoteAudioSource since it is reference counted
// in the message.
delete msg->pdata;
}
} // namespace webrtc