Index: content/renderer/media/webrtc_local_audio_track.cc |
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc |
deleted file mode 100644 |
index cb48668eaeccc09a670c0a150b5d439e8d515778..0000000000000000000000000000000000000000 |
--- a/content/renderer/media/webrtc_local_audio_track.cc |
+++ /dev/null |
@@ -1,241 +0,0 @@ |
-// Copyright (c) 2012 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_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/webaudio_capturer_source.h" |
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
-#include "content/renderer/media/webrtc_audio_capturer.h" |
- |
-namespace content { |
- |
-WebRtcLocalAudioTrack::WebRtcLocalAudioTrack( |
- WebRtcLocalAudioTrackAdapter* adapter, |
- const scoped_refptr<WebRtcAudioCapturer>& capturer, |
- WebAudioCapturerSource* webaudio_source) |
- : MediaStreamAudioTrack(true), |
- adapter_(adapter), |
- capturer_(capturer), |
- webaudio_source_(webaudio_source) { |
- DCHECK(capturer.get() || webaudio_source); |
- signal_thread_checker_.DetachFromThread(); |
- |
- adapter_->Initialize(this); |
- |
- DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()"; |
-} |
- |
-WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()"; |
- // Users might not call Stop() on the track. |
- Stop(); |
-} |
- |
-media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- if (webaudio_source_.get()) { |
- return media::AudioParameters(); |
- } else { |
- return capturer_->GetOutputFormat(); |
- } |
-} |
- |
-void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus, |
- base::TimeTicks estimated_capture_time, |
- bool force_report_nonzero_energy) { |
- DCHECK(capture_thread_checker_.CalledOnValidThread()); |
- DCHECK(!estimated_capture_time.is_null()); |
- |
- // Calculate the signal level regardless of whether the track is disabled or |
- // enabled. If |force_report_nonzero_energy| is true, |audio_bus| contains |
- // post-processed data that may be all zeros even though the signal contained |
- // energy before the processing. In this case, report nonzero energy even if |
- // the energy of the data in |audio_bus| is zero. |
- const float minimum_signal_level = |
- force_report_nonzero_energy ? 1.0f / std::numeric_limits<int16_t>::max() |
- : 0.0f; |
- const float signal_level = std::max( |
- minimum_signal_level, |
- std::min(1.0f, level_calculator_->Calculate(audio_bus))); |
- const int signal_level_as_pcm16 = |
- static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() + |
- 0.5f /* rounding to nearest int */); |
- adapter_->SetSignalLevel(signal_level_as_pcm16); |
- |
- scoped_refptr<WebRtcAudioCapturer> capturer; |
- SinkList::ItemList sinks; |
- SinkList::ItemList sinks_to_notify_format; |
- { |
- base::AutoLock auto_lock(lock_); |
- capturer = capturer_; |
- 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. |
- 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 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()); |
- |
- audio_parameters_ = params; |
- level_calculator_.reset(new MediaStreamAudioLevelCalculator()); |
- |
- base::AutoLock auto_lock(lock_); |
- // Remember to notify all sinks of the new format. |
- sinks_.TagAll(); |
-} |
- |
-void WebRtcLocalAudioTrack::SetAudioProcessor( |
- const scoped_refptr<MediaStreamAudioProcessor>& processor) { |
- // if the |processor| does not have audio processing, which can happen if |
- // kDisableAudioTrackProcessing is set set or all the constraints in |
- // the |processor| are turned off. In such case, we pass NULL to the |
- // adapter to indicate that no stats can be gotten from the processor. |
- adapter_->SetAudioProcessor(processor->has_audio_processing() ? |
- processor : NULL); |
-} |
- |
-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::Start() { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- DVLOG(1) << "WebRtcLocalAudioTrack::Start()"; |
- if (webaudio_source_.get()) { |
- // If the track is hooking up with WebAudio, do NOT add the track to the |
- // capturer as its sink otherwise two streams in different clock will be |
- // pushed through the same track. |
- webaudio_source_->Start(this); |
- } else if (capturer_.get()) { |
- capturer_->AddTrack(this); |
- } |
- |
- SinkList::ItemList sinks; |
- { |
- base::AutoLock auto_lock(lock_); |
- sinks = sinks_.Items(); |
- } |
- for (SinkList::ItemList::const_iterator it = sinks.begin(); |
- it != sinks.end(); |
- ++it) { |
- (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateLive); |
- } |
-} |
- |
-void WebRtcLocalAudioTrack::SetEnabled(bool enabled) { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- if (adapter_.get()) |
- adapter_->set_enabled(enabled); |
-} |
- |
-void WebRtcLocalAudioTrack::Stop() { |
- DCHECK(main_render_thread_checker_.CalledOnValidThread()); |
- DVLOG(1) << "WebRtcLocalAudioTrack::Stop()"; |
- if (!capturer_.get() && !webaudio_source_.get()) |
- return; |
- |
- if (webaudio_source_.get()) { |
- // Called Stop() on the |webaudio_source_| explicitly so that |
- // |webaudio_source_| won't push more data to the track anymore. |
- // Also note that the track is not registered as a sink to the |capturer_| |
- // in such case and no need to call RemoveTrack(). |
- webaudio_source_->Stop(); |
- } else { |
- // It is necessary to call RemoveTrack on the |capturer_| to avoid getting |
- // audio callback after Stop(). |
- capturer_->RemoveTrack(this); |
- } |
- |
- // Protect the pointers using the lock when accessing |sinks_| and |
- // setting the |capturer_| to NULL. |
- SinkList::ItemList sinks; |
- { |
- base::AutoLock auto_lock(lock_); |
- sinks = sinks_.Items(); |
- sinks_.Clear(); |
- webaudio_source_ = NULL; |
- capturer_ = NULL; |
- } |
- |
- 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 |