Index: content/renderer/media/webrtc/processed_local_audio_track.cc |
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc/processed_local_audio_track.cc |
similarity index 14% |
rename from content/renderer/media/webrtc_local_audio_track.cc |
rename to content/renderer/media/webrtc/processed_local_audio_track.cc |
index 53c55b2d0d4c45a7c50c331b547c616dd4f717e2..dad66619b03f5d409c4d69d9e831f60926de4fd2 100644 |
--- a/content/renderer/media/webrtc_local_audio_track.cc |
+++ b/content/renderer/media/webrtc/processed_local_audio_track.cc |
@@ -2,168 +2,60 @@ |
// 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_local_audio_track.h" |
+#include "content/renderer/media/webrtc/processed_local_audio_track.h" |
-#include <stdint.h> |
- |
-#include <limits> |
- |
-#include "content/public/renderer/media_stream_audio_sink.h" |
-#include "content/renderer/media/media_stream_audio_level_calculator.h" |
-#include "content/renderer/media/media_stream_audio_processor.h" |
-#include "content/renderer/media/media_stream_audio_sink_owner.h" |
-#include "content/renderer/media/media_stream_audio_track_sink.h" |
#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
namespace content { |
-WebRtcLocalAudioTrack::WebRtcLocalAudioTrack( |
+namespace { |
+// Used as an identifier for ProcessedLocalAudioTrack::From(). |
+void* const kClassIdentifier = const_cast<void**>(&kClassIdentifier); |
+} // namespace |
+ |
+ProcessedLocalAudioTrack::ProcessedLocalAudioTrack( |
scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter) |
: MediaStreamAudioTrack(true), adapter_(std::move(adapter)) { |
- signal_thread_checker_.DetachFromThread(); |
- DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()"; |
+ DCHECK(adapter_); |
+ adapter_->SetMediaStreamAudioTrack(MediaStreamAudioTrack::GetWeakPtr()); |
- adapter_->Initialize(this); |
+ DVLOG(1) << "ProcessedLocalAudioTrack::ProcessedLocalAudioTrack()"; |
} |
-WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() { |
+ProcessedLocalAudioTrack::~ProcessedLocalAudioTrack() { |
DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()"; |
+ DVLOG(1) << "ProcessedLocalAudioTrack::~ProcessedLocalAudioTrack()"; |
// Ensure the track is stopped. |
MediaStreamAudioTrack::Stop(); |
} |
-media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- base::AutoLock auto_lock(lock_); |
- return audio_parameters_; |
+// static |
+ProcessedLocalAudioTrack* ProcessedLocalAudioTrack::From( |
+ MediaStreamAudioTrack* track) { |
+ if (track && track->GetClassIdentifier() == kClassIdentifier) |
+ return static_cast<ProcessedLocalAudioTrack*>(track); |
+ return nullptr; |
} |
-void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus, |
- base::TimeTicks estimated_capture_time) { |
- DCHECK(capture_thread_checker_.CalledOnValidThread()); |
- DCHECK(!estimated_capture_time.is_null()); |
- |
- SinkList::ItemList sinks; |
- SinkList::ItemList sinks_to_notify_format; |
- { |
- base::AutoLock auto_lock(lock_); |
- sinks = sinks_.Items(); |
- sinks_.RetrieveAndClearTags(&sinks_to_notify_format); |
- } |
- |
- // Notify the tracks on when the format changes. This will do nothing if |
- // |sinks_to_notify_format| is empty. Note that accessing |audio_parameters_| |
- // without holding the |lock_| is valid since |audio_parameters_| is only |
- // changed on the current thread. |
- for (const auto& sink : sinks_to_notify_format) |
- sink->OnSetFormat(audio_parameters_); |
- |
- // Feed the data to the sinks. |
- // TODO(jiayl): we should not pass the real audio data down if the track is |
- // disabled. This is currently done so to feed input to WebRTC typing |
- // detection and should be changed when audio processing is moved from |
- // WebRTC to the track. |
- for (const auto& sink : sinks) |
- sink->OnData(audio_bus, estimated_capture_time); |
+void* ProcessedLocalAudioTrack::GetClassIdentifier() const { |
+ return kClassIdentifier; |
} |
-void WebRtcLocalAudioTrack::OnSetFormat( |
- const media::AudioParameters& params) { |
- DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()"; |
- // If the source is restarted, we might have changed to another capture |
- // thread. |
- capture_thread_checker_.DetachFromThread(); |
- DCHECK(capture_thread_checker_.CalledOnValidThread()); |
- |
- base::AutoLock auto_lock(lock_); |
- audio_parameters_ = params; |
- // Remember to notify all sinks of the new format. |
- sinks_.TagAll(); |
-} |
- |
-void WebRtcLocalAudioTrack::SetLevel( |
+void ProcessedLocalAudioTrack::SetLevel( |
scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) { |
adapter_->SetLevel(std::move(level)); |
} |
-void WebRtcLocalAudioTrack::SetAudioProcessor( |
+void ProcessedLocalAudioTrack::SetAudioProcessor( |
scoped_refptr<MediaStreamAudioProcessor> processor) { |
adapter_->SetAudioProcessor(std::move(processor)); |
} |
-void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) { |
- // This method is called from webrtc, on the signaling thread, when the local |
- // description is set and from the main thread from WebMediaPlayerMS::load |
- // (via WebRtcLocalAudioRenderer::Start). |
- DCHECK(main_render_thread_checker_.CalledOnValidThread() || |
- signal_thread_checker_.CalledOnValidThread()); |
- DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()"; |
- base::AutoLock auto_lock(lock_); |
- |
- // Verify that |sink| is not already added to the list. |
- DCHECK(!sinks_.Contains( |
- MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink))); |
- |
- // Create (and add to the list) a new MediaStreamAudioTrackSink |
- // which owns the |sink| and delagates all calls to the |
- // MediaStreamAudioSink interface. It will be tagged in the list, so |
- // we remember to call OnSetFormat() on the new sink. |
- scoped_refptr<MediaStreamAudioTrackSink> sink_owner( |
- new MediaStreamAudioSinkOwner(sink)); |
- sinks_.AddAndTag(sink_owner.get()); |
-} |
- |
-void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) { |
- // See AddSink for additional context. When local audio is stopped from |
- // webrtc, we'll be called here on the signaling thread. |
- DCHECK(main_render_thread_checker_.CalledOnValidThread() || |
- signal_thread_checker_.CalledOnValidThread()); |
- DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()"; |
- |
- scoped_refptr<MediaStreamAudioTrackSink> removed_item; |
- { |
- base::AutoLock auto_lock(lock_); |
- removed_item = sinks_.Remove( |
- MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)); |
- } |
- |
- // Clear the delegate to ensure that no more capture callbacks will |
- // be sent to this sink. Also avoids a possible crash which can happen |
- // if this method is called while capturing is active. |
- if (removed_item.get()) |
- removed_item->Reset(); |
-} |
- |
-void WebRtcLocalAudioTrack::SetEnabled(bool enabled) { |
+void ProcessedLocalAudioTrack::SetEnabled(bool enabled) { |
DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
+ MediaStreamAudioTrack::SetEnabled(enabled); |
if (adapter_.get()) |
adapter_->set_enabled(enabled); |
} |
-void WebRtcLocalAudioTrack::OnStop() { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- DVLOG(1) << "WebRtcLocalAudioTrack::OnStop()"; |
- |
- // Protect the pointers using the lock when accessing |sinks_|. |
- SinkList::ItemList sinks; |
- { |
- base::AutoLock auto_lock(lock_); |
- sinks = sinks_.Items(); |
- sinks_.Clear(); |
- } |
- |
- for (SinkList::ItemList::const_iterator it = sinks.begin(); |
- it != sinks.end(); |
- ++it){ |
- (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded); |
- (*it)->Reset(); |
- } |
-} |
- |
-webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- return adapter_.get(); |
-} |
- |
} // namespace content |