| 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 53c55b2d0d4c45a7c50c331b547c616dd4f717e2..0000000000000000000000000000000000000000
|
| --- a/content/renderer/media/webrtc_local_audio_track.cc
|
| +++ /dev/null
|
| @@ -1,169 +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/webrtc/webrtc_local_audio_track_adapter.h"
|
| -
|
| -namespace content {
|
| -
|
| -WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
|
| - scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter)
|
| - : MediaStreamAudioTrack(true), adapter_(std::move(adapter)) {
|
| - signal_thread_checker_.DetachFromThread();
|
| - DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
|
| -
|
| - adapter_->Initialize(this);
|
| -}
|
| -
|
| -WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
|
| - // 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_;
|
| -}
|
| -
|
| -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_);
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| - 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.
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| - // 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.
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| - 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());
|
| -
|
| - base::AutoLock auto_lock(lock_);
|
| - audio_parameters_ = params;
|
| - // Remember to notify all sinks of the new format.
|
| - sinks_.TagAll();
|
| -}
|
| -
|
| -void WebRtcLocalAudioTrack::SetLevel(
|
| - scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
|
| - adapter_->SetLevel(std::move(level));
|
| -}
|
| -
|
| -void WebRtcLocalAudioTrack::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
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| - // 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.
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| - scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
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| - 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) {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - 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
|
|
|