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// Copyright 2016 The Chromium Authors
// 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/peerconnection/webrtc_audio_sink.h"
#include <algorithm>
#include <limits>
#include "base/check_op.h"
#include "base/containers/contains.h"
#include "base/functional/callback_helpers.h"
#include "base/location.h"
#include "base/memory/scoped_refptr.h"
#include "base/strings/stringprintf.h"
#include "base/strings/to_string.h"
#include "base/task/single_thread_task_runner.h"
#include "base/trace_event/trace_event.h"
#include "media/base/audio_timestamp_helper.h"
#include "third_party/blink/public/common/features.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h"
#include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h"
#include "third_party/webrtc/rtc_base/ref_counted_object.h"
#include "third_party/webrtc/rtc_base/time_utils.h"
namespace {
void SendLogMessage(const std::string& message) {
blink::WebRtcLogMessage("WRAS::" + message);
}
} // namespace
namespace WTF {
template <>
struct CrossThreadCopier<scoped_refptr<webrtc::AudioProcessorInterface>>
: public CrossThreadCopierByValuePassThrough<
scoped_refptr<webrtc::AudioProcessorInterface>> {
STATIC_ONLY(CrossThreadCopier);
};
template <>
struct CrossThreadCopier<scoped_refptr<blink::WebRtcAudioSink::Adapter>>
: public CrossThreadCopierPassThrough<
scoped_refptr<blink::WebRtcAudioSink::Adapter>> {
STATIC_ONLY(CrossThreadCopier);
};
} // namespace WTF
namespace blink {
WebRtcAudioSink::WebRtcAudioSink(
const std::string& label,
scoped_refptr<webrtc::AudioSourceInterface> track_source,
scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner,
scoped_refptr<base::SingleThreadTaskRunner> main_task_runner)
: adapter_(new webrtc::RefCountedObject<Adapter>(
label,
std::move(track_source),
std::move(signaling_task_runner),
std::move(main_task_runner))),
fifo_(ConvertToBaseRepeatingCallback(
CrossThreadBindRepeating(&WebRtcAudioSink::DeliverRebufferedAudio,
CrossThreadUnretained(this)))),
num_preferred_channels_(-1) {
SendLogMessage(base::StringPrintf("WebRtcAudioSink({label=%s})",
adapter_->label().c_str()));
}
WebRtcAudioSink::~WebRtcAudioSink() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
SendLogMessage(base::StringPrintf("~WebRtcAudioSink([label=%s])",
adapter_->label().c_str()));
}
void WebRtcAudioSink::SetAudioProcessor(
scoped_refptr<webrtc::AudioProcessorInterface> processor) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
DCHECK(processor.get());
adapter_->set_processor(std::move(processor));
}
void WebRtcAudioSink::SetLevel(
scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
DCHECK(level.get());
adapter_->set_level(std::move(level));
}
void WebRtcAudioSink::OnEnabledChanged(bool enabled) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
SendLogMessage(base::StringPrintf("OnEnabledChanged([label=%s] {enabled=%s})",
adapter_->label().c_str(),
base::ToString(enabled).c_str()));
PostCrossThreadTask(
*adapter_->signaling_task_runner(), FROM_HERE,
CrossThreadBindOnce(
base::IgnoreResult(&WebRtcAudioSink::Adapter::set_enabled), adapter_,
enabled));
}
void WebRtcAudioSink::OnData(const media::AudioBus& audio_bus,
base::TimeTicks estimated_capture_time) {
// No thread check: OnData might be called on different threads (but not
// concurrently).
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("mediastream"),
"WebRtcAudioSink::OnData", "this", static_cast<void*>(this),
"frames", audio_bus.frames());
// TODO(crbug.com/1054769): Better to let |fifo_| handle the estimated capture
// time and let it return a corrected interpolated capture time to
// DeliverRebufferedAudio(). Current, similar treatment is used at different
// places where |AudioPushFifo| is applied. So a update to |AudioPushFifo|
// will be a joint effort, and should be carefully carried out.
last_estimated_capture_time_ = estimated_capture_time;
if (base::FeatureList::IsEnabled(
features::kWebRtcAudioSinkUseTimestampAligner)) {
adapter_->UpdateTimestampAligner(estimated_capture_time);
}
// The following will result in zero, one, or multiple synchronous calls to
// DeliverRebufferedAudio().
fifo_.Push(audio_bus);
}
void WebRtcAudioSink::OnSetFormat(const media::AudioParameters& params) {
CHECK(params.IsValid());
SendLogMessage(base::StringPrintf("OnSetFormat([label=%s] {params=[%s]})",
adapter_->label().c_str(),
params.AsHumanReadableString().c_str()));
params_ = params;
// Make sure that our params always reflect a buffer size of 10ms.
params_.set_frames_per_buffer(params_.sample_rate() / 100);
fifo_.Reset(params_.frames_per_buffer());
const int num_pcm16_data_elements =
params_.frames_per_buffer() * params_.channels();
interleaved_data_.reset(new int16_t[num_pcm16_data_elements]);
}
void WebRtcAudioSink::DeliverRebufferedAudio(const media::AudioBus& audio_bus,
int frame_delay) {
DCHECK(params_.IsValid());
TRACE_EVENT1("audio", "WebRtcAudioSink::DeliverRebufferedAudio", "frames",
audio_bus.frames());
// TODO(henrika): Remove this conversion once the interface in libjingle
// supports float vectors.
static_assert(sizeof(interleaved_data_[0]) == 2,
"ToInterleaved expects 2 bytes.");
audio_bus.ToInterleaved<media::SignedInt16SampleTypeTraits>(
audio_bus.frames(), interleaved_data_.get());
const base::TimeTicks estimated_capture_time =
last_estimated_capture_time_ + media::AudioTimestampHelper::FramesToTime(
frame_delay, params_.sample_rate());
num_preferred_channels_ = adapter_->DeliverPCMToWebRtcSinks(
interleaved_data_.get(), params_.sample_rate(), audio_bus.channels(),
audio_bus.frames(), estimated_capture_time);
}
namespace {
void DereferenceOnMainThread(
scoped_refptr<webrtc::AudioProcessorInterface> processor) {
// The ref count was artificially increased before posting the task. Decrease
// it again to ensure that the processor is destroyed when the scoped_refptr
// goes out of scope.
processor->Release();
}
} // namespace
WebRtcAudioSink::Adapter::Adapter(
const std::string& label,
scoped_refptr<webrtc::AudioSourceInterface> source,
scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner,
scoped_refptr<base::SingleThreadTaskRunner> main_task_runner)
: webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
label_(label),
source_(std::move(source)),
signaling_task_runner_(std::move(signaling_task_runner)),
main_task_runner_(std::move(main_task_runner)) {
DCHECK(signaling_task_runner_);
DCHECK(main_task_runner_);
SendLogMessage(
base::StringPrintf("Adapter::Adapter({label=%s})", label_.c_str()));
}
WebRtcAudioSink::Adapter::~Adapter() {
SendLogMessage(
base::StringPrintf("Adapter::~Adapter([label=%s])", label_.c_str()));
if (audio_processor_) {
// Artificially increase the ref count of audio_processor_ before posting it
// to the main thread to be destroyed. If the post succeeds, it will be
// destroyed on the main thread as intended. If the post fails, the ref
// count will remain at 1, leaking the processor. This is preferred to
// destroying it on the wrong thread, which causes a crash.
audio_processor_->AddRef();
auto* possible_leak = audio_processor_.get();
if (!PostCrossThreadTask(
*main_task_runner_.get(), FROM_HERE,
CrossThreadBindOnce(&DereferenceOnMainThread,
std::move(audio_processor_)))) {
DVLOG(1) << __func__
<< " Intentionally leaking audio_processor_ due to failed "
"PostCrossThreadTask: "
<< possible_leak;
}
}
}
int WebRtcAudioSink::Adapter::DeliverPCMToWebRtcSinks(
const int16_t* audio_data,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
base::TimeTicks estimated_capture_time) {
base::AutoLock auto_lock(lock_);
int64_t capture_timestamp_ms =
estimated_capture_time.since_origin().InMilliseconds();
if (base::FeatureList::IsEnabled(
features::kWebRtcAudioSinkUseTimestampAligner)) {
// This use |timestamp_aligner_| to transform |estimated_capture_timestamp|
// to webrtc::TimeMicros(). See the comment at UpdateTimestampAligner() for
// more details.
capture_timestamp_ms =
timestamp_aligner_.TranslateTimestamp(
estimated_capture_time.since_origin().InMicroseconds()) /
webrtc::kNumMicrosecsPerMillisec;
}
int num_preferred_channels = -1;
for (webrtc::AudioTrackSinkInterface* sink : sinks_) {
sink->OnData(audio_data, sizeof(int16_t) * 8, sample_rate,
number_of_channels, number_of_frames, capture_timestamp_ms);
num_preferred_channels =
std::max(num_preferred_channels, sink->NumPreferredChannels());
}
return num_preferred_channels;
}
std::string WebRtcAudioSink::Adapter::kind() const {
return webrtc::MediaStreamTrackInterface::kAudioKind;
}
bool WebRtcAudioSink::Adapter::set_enabled(bool enable) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence());
SendLogMessage(
base::StringPrintf("Adapter::set_enabled([label=%s] {enable=%s})",
label_.c_str(), base::ToString(enable).c_str()));
return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::set_enabled(
enable);
}
void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence());
DCHECK(sink);
SendLogMessage(
base::StringPrintf("Adapter::AddSink({label=%s})", label_.c_str()));
base::AutoLock auto_lock(lock_);
DCHECK(!base::Contains(sinks_, sink));
sinks_.push_back(sink);
}
void WebRtcAudioSink::Adapter::RemoveSink(
webrtc::AudioTrackSinkInterface* sink) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence());
SendLogMessage(
base::StringPrintf("Adapter::RemoveSink([label=%s])", label_.c_str()));
base::AutoLock auto_lock(lock_);
auto it = std::ranges::find(sinks_, sink);
if (it != sinks_.end())
sinks_.erase(it);
}
bool WebRtcAudioSink::Adapter::GetSignalLevel(int* level) {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence());
// |level_| is only set once, so it's safe to read without first acquiring a
// mutex.
if (!level_)
return false;
const float signal_level = level_->GetCurrent();
DCHECK_GE(signal_level, 0.0f);
DCHECK_LE(signal_level, 1.0f);
// Convert from float in range [0.0,1.0] to an int in range [0,32767].
*level = static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
0.5f /* rounding to nearest int */);
// TODO(crbug/1073391): possibly log the signal level but first check the
// calling frequency of this method to avoid creating too much data.
return true;
}
webrtc::scoped_refptr<webrtc::AudioProcessorInterface>
WebRtcAudioSink::Adapter::GetAudioProcessor() {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence());
return webrtc::scoped_refptr<webrtc::AudioProcessorInterface>(
audio_processor_.get());
}
webrtc::AudioSourceInterface* WebRtcAudioSink::Adapter::GetSource() const {
DCHECK(!signaling_task_runner_ ||
signaling_task_runner_->RunsTasksInCurrentSequence());
return source_.get();
}
void WebRtcAudioSink::Adapter::UpdateTimestampAligner(
base::TimeTicks capture_time) {
// The |timestamp_aligner_| stamps an audio frame as if it is captured 'now',
// taking webrtc::TimeMicros as the reference clock. It does not provide the
// time that the frame was originally captured, Using |timestamp_aligner_|
// rather than calling webrtc::TimeMicros is to take the advantage that it
// aligns its output timestamps such that the time spacing in the
// |capture_time| is maintained.
timestamp_aligner_.TranslateTimestamp(
capture_time.since_origin().InMicroseconds(), webrtc::TimeMicros());
}
} // namespace blink
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