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Unified Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc

Issue 1780653002: Revert of MediaStream audio object graph untangling and clean-ups. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 9 months ago
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Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
index db41413d9ada657e75d195cf4d491d56a90e8f43..20a3969d11312d81c0b99354bc08c3f476666884 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
@@ -23,31 +23,34 @@
webrtc::AudioSourceInterface* track_source) {
// TODO(tommi): Change this so that the signaling thread is one of the
// parameters to this method.
- scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner;
+ scoped_refptr<base::SingleThreadTaskRunner> signaling_thread;
RenderThreadImpl* current = RenderThreadImpl::current();
if (current) {
PeerConnectionDependencyFactory* pc_factory =
current->GetPeerConnectionDependencyFactory();
- signaling_task_runner = pc_factory->GetWebRtcSignalingThread();
- CHECK(signaling_task_runner);
- } else {
- LOG(WARNING) << "Assuming single-threaded operation for unit test.";
+ signaling_thread = pc_factory->GetWebRtcSignalingThread();
}
+
+ LOG_IF(ERROR, !signaling_thread.get()) << "No signaling thread!";
rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter =
new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>(
- label, track_source, std::move(signaling_task_runner));
+ label, track_source, signaling_thread);
return adapter;
}
WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
const std::string& label,
webrtc::AudioSourceInterface* track_source,
- scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
+ const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread)
: webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
owner_(NULL),
track_source_(track_source),
- signaling_task_runner_(std::move(signaling_task_runner)) {}
+ signaling_thread_(signaling_thread),
+ signal_level_(0) {
+ signaling_thread_checker_.DetachFromThread();
+ capture_thread_.DetachFromThread();
+}
WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
}
@@ -59,17 +62,14 @@
}
void WebRtcLocalAudioTrackAdapter::SetAudioProcessor(
- scoped_refptr<MediaStreamAudioProcessor> processor) {
- DCHECK(processor.get());
- DCHECK(!audio_processor_);
- audio_processor_ = std::move(processor);
-}
-
-void WebRtcLocalAudioTrackAdapter::SetLevel(
- scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
- DCHECK(level.get());
- DCHECK(!level_);
- level_ = std::move(level);
+ const scoped_refptr<MediaStreamAudioProcessor>& processor) {
+ // SetAudioProcessor will be called when a new capture thread has been
+ // initialized, so we need to detach from any current capture thread we're
+ // checking and attach to the current one.
+ capture_thread_.DetachFromThread();
+ DCHECK(capture_thread_.CalledOnValidThread());
+ base::AutoLock auto_lock(lock_);
+ audio_processor_ = processor;
}
std::string WebRtcLocalAudioTrackAdapter::kind() const {
@@ -79,9 +79,8 @@
bool WebRtcLocalAudioTrackAdapter::set_enabled(bool enable) {
// If we're not called on the signaling thread, we need to post a task to
// change the state on the correct thread.
- if (signaling_task_runner_ &&
- !signaling_task_runner_->BelongsToCurrentThread()) {
- signaling_task_runner_->PostTask(FROM_HERE,
+ if (signaling_thread_.get() && !signaling_thread_->BelongsToCurrentThread()) {
+ signaling_thread_->PostTask(FROM_HERE,
base::Bind(
base::IgnoreResult(&WebRtcLocalAudioTrackAdapter::set_enabled),
this, enable));
@@ -94,8 +93,7 @@
void WebRtcLocalAudioTrackAdapter::AddSink(
webrtc::AudioTrackSinkInterface* sink) {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
+ DCHECK(signaling_thread_checker_.CalledOnValidThread());
DCHECK(sink);
#ifndef NDEBUG
// Verify that |sink| has not been added.
@@ -114,8 +112,7 @@
void WebRtcLocalAudioTrackAdapter::RemoveSink(
webrtc::AudioTrackSinkInterface* sink) {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
+ DCHECK(signaling_thread_checker_.CalledOnValidThread());
DCHECK(sink);
for (ScopedVector<WebRtcAudioSinkAdapter>::iterator it =
sink_adapters_.begin();
@@ -129,32 +126,28 @@
}
bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
+ DCHECK(signaling_thread_checker_.CalledOnValidThread());
- // |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 */);
+ base::AutoLock auto_lock(lock_);
+ *level = signal_level_;
return true;
}
rtc::scoped_refptr<webrtc::AudioProcessorInterface>
WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
+ DCHECK(signaling_thread_checker_.CalledOnValidThread());
+ base::AutoLock auto_lock(lock_);
return audio_processor_.get();
}
+void WebRtcLocalAudioTrackAdapter::SetSignalLevel(int signal_level) {
+ DCHECK(capture_thread_.CalledOnValidThread());
+ base::AutoLock auto_lock(lock_);
+ signal_level_ = signal_level;
+}
+
webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
+ DCHECK(signaling_thread_checker_.CalledOnValidThread());
return track_source_;
}

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