Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(3302)

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

Issue 1834323002: MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: REBASE + Workaround to ensure MediaStreamAudioProcessor is destroyed on the main thread. Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
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
deleted file mode 100644
index c86881b07a90eed64bca82048846e137536a2b21..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
+++ /dev/null
@@ -1,161 +0,0 @@
-// Copyright 2014 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 "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-
-#include "base/location.h"
-#include "base/logging.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
-#include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-#include "content/renderer/render_thread_impl.h"
-#include "third_party/webrtc/api/mediastreaminterface.h"
-
-namespace content {
-
-static const char kAudioTrackKind[] = "audio";
-
-scoped_refptr<WebRtcLocalAudioTrackAdapter>
-WebRtcLocalAudioTrackAdapter::Create(
- const std::string& label,
- 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;
- RenderThreadImpl* current = RenderThreadImpl::current();
- if (current) {
- PeerConnectionDependencyFactory* pc_factory =
- current->GetPeerConnectionDependencyFactory();
- signaling_task_runner = pc_factory->GetWebRtcSignalingThread();
- LOG_IF(ERROR, !signaling_task_runner) << "No signaling thread!";
- } else {
- LOG(WARNING) << "Assuming single-threaded operation for unit test.";
- }
-
- rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>* adapter =
- new rtc::RefCountedObject<WebRtcLocalAudioTrackAdapter>(
- label, track_source, std::move(signaling_task_runner));
- return adapter;
-}
-
-WebRtcLocalAudioTrackAdapter::WebRtcLocalAudioTrackAdapter(
- const std::string& label,
- webrtc::AudioSourceInterface* track_source,
- scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner)
- : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(label),
- owner_(NULL),
- track_source_(track_source),
- signaling_task_runner_(std::move(signaling_task_runner)) {}
-
-WebRtcLocalAudioTrackAdapter::~WebRtcLocalAudioTrackAdapter() {
-}
-
-void WebRtcLocalAudioTrackAdapter::Initialize(WebRtcLocalAudioTrack* owner) {
- DCHECK(!owner_);
- DCHECK(owner);
- owner_ = owner;
-}
-
-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);
-}
-
-std::string WebRtcLocalAudioTrackAdapter::kind() const {
- return kAudioTrackKind;
-}
-
-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,
- base::Bind(
- base::IgnoreResult(&WebRtcLocalAudioTrackAdapter::set_enabled),
- this, enable));
- return true;
- }
-
- return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
- set_enabled(enable);
-}
-
-void WebRtcLocalAudioTrackAdapter::AddSink(
- webrtc::AudioTrackSinkInterface* sink) {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
- DCHECK(sink);
-#ifndef NDEBUG
- // Verify that |sink| has not been added.
- for (ScopedVector<WebRtcAudioSinkAdapter>::const_iterator it =
- sink_adapters_.begin();
- it != sink_adapters_.end(); ++it) {
- DCHECK(!(*it)->IsEqual(sink));
- }
-#endif
-
- std::unique_ptr<WebRtcAudioSinkAdapter> adapter(
- new WebRtcAudioSinkAdapter(sink));
- owner_->AddSink(adapter.get());
- sink_adapters_.push_back(adapter.release());
-}
-
-void WebRtcLocalAudioTrackAdapter::RemoveSink(
- webrtc::AudioTrackSinkInterface* sink) {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
- DCHECK(sink);
- for (ScopedVector<WebRtcAudioSinkAdapter>::iterator it =
- sink_adapters_.begin();
- it != sink_adapters_.end(); ++it) {
- if ((*it)->IsEqual(sink)) {
- owner_->RemoveSink(*it);
- sink_adapters_.erase(it);
- return;
- }
- }
-}
-
-bool WebRtcLocalAudioTrackAdapter::GetSignalLevel(int* level) {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
-
- // |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 */);
- return true;
-}
-
-rtc::scoped_refptr<webrtc::AudioProcessorInterface>
-WebRtcLocalAudioTrackAdapter::GetAudioProcessor() {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
- return audio_processor_.get();
-}
-
-webrtc::AudioSourceInterface* WebRtcLocalAudioTrackAdapter::GetSource() const {
- DCHECK(!signaling_task_runner_ ||
- signaling_task_runner_->RunsTasksOnCurrentThread());
- return track_source_;
-}
-
-} // namespace content

Powered by Google App Engine
This is Rietveld 408576698