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

Issue 1966043006: Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 7 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
new file mode 100644
index 0000000000000000000000000000000000000000..c86881b07a90eed64bca82048846e137536a2b21
--- /dev/null
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc
@@ -0,0 +1,161 @@
+// 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

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