move the APM to chrome.

content/renderer/media/webrtc_audio_capturer.cc

 // 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_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 45 matching lines @@
   }
 
   void SetCaptureFormat(const media::AudioParameters& params) {
     base::AutoLock lock(lock_);
     if (delegate_)
       delegate_->SetCaptureFormat(params);
   }
 
   void Reset() {
     base::AutoLock lock(lock_);
+    if (delegate_)
+      delegate_->Stop();
     delegate_ = NULL;
   }
 
   // Wrapper which allows to use std::find_if() when adding and removing
   // sinks to/from the list.
   struct TrackWrapper {
     TrackWrapper(WebRtcLocalAudioTrack* track) : track_(track) {}
     bool operator()(
         const scoped_refptr<WebRtcAudioCapturer::TrackOwner>& owner) {
       return owner->IsEqual(track_);
@@ skipping 18 matching lines @@
 
   DISALLOW_COPY_AND_ASSIGN(TrackOwner);
 };
 
 // static
 scoped_refptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer() {
   scoped_refptr<WebRtcAudioCapturer> capturer = new WebRtcAudioCapturer();
   return capturer;
 }
 
-void WebRtcAudioCapturer::Reconfigure(int sample_rate,
-                                      media::ChannelLayout channel_layout) {
+void WebRtcAudioCapturer::Reconfigure(
+    int sample_rate,
+    media::ChannelLayout channel_layout) {
   DCHECK(thread_checker_.CalledOnValidThread());
   int buffer_size = GetBufferSize(sample_rate);
   DVLOG(1) << "Using WebRTC input buffer size: " << buffer_size;
 
   media::AudioParameters::Format format =
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY;
 
   // bits_per_sample is always 16 for now.
   int bits_per_sample = 16;
   media::AudioParameters params(format, channel_layout, sample_rate,
                                 bits_per_sample, buffer_size);
+  DCHECK(params.IsValid());
 
   TrackList tracks;
   {
     base::AutoLock auto_lock(lock_);
     tracks = tracks_;
     params_ = params;
   }
 
   // Tell all audio_tracks which format we use.
   for (TrackList::const_iterator it = tracks.begin();
        it != tracks.end(); ++it)
     (*it)->SetCaptureFormat(params);
 }
 
-bool WebRtcAudioCapturer::Initialize(int render_view_id,
-                                     media::ChannelLayout channel_layout,
-                                     int sample_rate,
-                                     int buffer_size,
-                                     int session_id,
-                                     const std::string& device_id,
-                                     int paired_output_sample_rate,
-                                     int paired_output_frames_per_buffer) {
+bool WebRtcAudioCapturer::Initialize(
+    int render_view_id,
+    media::ChannelLayout channel_layout,
+    int sample_rate,
+    int buffer_size,
+    int session_id,
+    const std::string& device_id,
+    int paired_output_sample_rate,
+    int paired_output_frames_per_buffer) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK_GE(render_view_id, 0);
   DVLOG(1) << "WebRtcAudioCapturer::Initialize()";
 
   DVLOG(1) << "Audio input hardware channel layout: " << channel_layout;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout",
                             channel_layout, media::CHANNEL_LAYOUT_MAX);
 
   render_view_id_ = render_view_id;
   session_id_ = session_id;
@@ skipping 31 matching lines @@
                 &kValidInputRates[0] + arraysize(kValidInputRates),
                 sample_rate) ==
           &kValidInputRates[arraysize(kValidInputRates)]) {
     DLOG(ERROR) << sample_rate << " is not a supported input rate.";
     return false;
   }
 
   // Create and configure the default audio capturing source. The |source_|
   // will be overwritten if an external client later calls SetCapturerSource()
   // providing an alternative media::AudioCapturerSource.
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
-                    channel_layout,
-                    static_cast<float>(sample_rate));
+  InitializeCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
+                           channel_layout,
+                           static_cast<float>(sample_rate));
 
   return true;
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer()
     : source_(NULL),
       running_(false),
       agc_is_enabled_(false),
       render_view_id_(-1),
       hardware_buffer_size_(0),
       session_id_(0),
       volume_(0),
-      source_provider_(new WebRtcLocalAudioSourceProvider()),
       peer_connection_mode_(false),
       output_sample_rate_(0),
-      output_frames_per_buffer_(0) {
-  DCHECK(source_provider_.get());
+      output_frames_per_buffer_(0),
+      audio_delay_ms_(0),
+      key_pressed_(false) {
   DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
 }
 
 WebRtcAudioCapturer::~WebRtcAudioCapturer() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(tracks_.empty());
   DCHECK(!running_);
   DVLOG(1) << "WebRtcAudioCapturer::~WebRtcAudioCapturer()";
 }
 
 void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) {
   DCHECK(track);
   DVLOG(1) << "WebRtcAudioCapturer::AddTrack()";
 
   // Start the source if the first audio track is connected to the capturer.
   // Start() will do nothing if the capturer has already been started.
   Start();
 
   base::AutoLock auto_lock(lock_);
   // Verify that |track| is not already added to the list.
   DCHECK(std::find_if(tracks_.begin(), tracks_.end(),
                       TrackOwner::TrackWrapper(track)) == tracks_.end());
 
-  track->SetCaptureFormat(params_);
+  if (params_.IsValid())
+    track->SetCaptureFormat(params_);
+
   tracks_.push_back(new WebRtcAudioCapturer::TrackOwner(track));
 }
 
 void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   bool stop_source = false;
   {
     base::AutoLock auto_lock(lock_);
     // Get iterator to the first element for which WrapsSink(track) returns
     // true.
     TrackList::iterator it = std::find_if(
         tracks_.begin(), tracks_.end(), TrackOwner::TrackWrapper(track));
     if (it != tracks_.end()) {
       // 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.
       (*it)->Reset();
       tracks_.erase(it);
     }
 
     // Stop the source if the last audio track is going away.
     stop_source = tracks_.empty();
   }
 
   if (stop_source)
     Stop();
 }
 
-void WebRtcAudioCapturer::SetCapturerSource(
+void WebRtcAudioCapturer::InitializeCapturerSource(
     const scoped_refptr<media::AudioCapturerSource>& source,
     media::ChannelLayout channel_layout,
     float sample_rate) {
   DCHECK(thread_checker_.CalledOnValidThread());
-  DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << ","
-           << "sample_rate=" << sample_rate << ")";
+  DVLOG(1) << "InitializeCapturerSource(channel_layout=" << channel_layout
+           << "," << "sample_rate=" << sample_rate << ")";
   scoped_refptr<media::AudioCapturerSource> old_source;
   bool restart_source = false;
   {
     base::AutoLock auto_lock(lock_);
     if (source_.get() == source.get())
       return;
 
     source_.swap(old_source);
     source_ = source;
 
     // Reset the flag to allow starting the new source.
     restart_source = running_;
     running_ = false;
   }
 
   DVLOG(1) << "Switching to a new capture source.";
   if (old_source.get())
     old_source->Stop();
 
   // Dispatch the new parameters both to the sink(s) and to the new source.
   // The idea is to get rid of any dependency of the microphone parameters
   // which would normally be used by default.
   Reconfigure(sample_rate, channel_layout);
 
   // Make sure to grab the new parameters in case they were reconfigured.
   media::AudioParameters params = audio_parameters();
-  source_provider_->Initialize(params);
   if (source.get())
     source->Initialize(params, this, session_id_);
 
   if (restart_source)
     Start();
 }
 
 void WebRtcAudioCapturer::EnablePeerConnectionMode() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DVLOG(1) << "EnablePeerConnectionMode";
@@ skipping 13 matching lines @@
     render_view_id = render_view_id_;
   }
 
   // Do nothing if the current buffer size is the WebRtc native buffer size.
   media::AudioParameters params = audio_parameters();
   if (GetBufferSize(params.sample_rate()) == params.frames_per_buffer())
     return;
 
   // Create a new audio stream as source which will open the hardware using
   // WebRtc native buffer size.
-  SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
-                    params.channel_layout(),
-                    static_cast<float>(params.sample_rate()));
+  InitializeCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id),
+                           params.channel_layout(),
+                           static_cast<float>(params.sample_rate()));
 }
 
 void WebRtcAudioCapturer::Start() {
   DVLOG(1) << "WebRtcAudioCapturer::Start()";
   base::AutoLock auto_lock(lock_);
   if (running_)
     return;
 
   // Start the data source, i.e., start capturing data from the current source.
   // Note that, the source does not have to be a microphone.
   if (source_.get()) {
     // We need to set the AGC control before starting the stream.
     source_->SetAutomaticGainControl(agc_is_enabled_);
     source_->Start();
   }
 
   running_ = true;
 }
 
 void WebRtcAudioCapturer::Stop() {
   DVLOG(1) << "WebRtcAudioCapturer::Stop()";
   scoped_refptr<media::AudioCapturerSource> source;
+  TrackList tracks;
   {
     base::AutoLock auto_lock(lock_);
     if (!running_)
       return;
 
     source = source_;
+    tracks = tracks_;
     running_ = false;
   }
 
   if (source.get())
     source->Stop();
+
+  for (TrackList::const_iterator it = tracks.begin(); it != tracks.end(); ++it)
+    (*it)->Reset();
 }
 
 void WebRtcAudioCapturer::SetVolume(int volume) {
   DVLOG(1) << "WebRtcAudioCapturer::SetVolume()";
   DCHECK_LE(volume, MaxVolume());
   double normalized_volume = static_cast<double>(volume) / MaxVolume();
   base::AutoLock auto_lock(lock_);
   if (source_.get())
     source_->SetVolume(normalized_volume);
 }
@@ skipping 43 matching lines @@
       return;
 
     // Map internal volume range of [0.0, 1.0] into [0, 255] used by the
     // webrtc::VoiceEngine. webrtc::VoiceEngine will handle the case when the
     // volume is higher than 255.
     volume_ = static_cast<int>((volume * MaxVolume()) + 0.5);
     current_volume = volume_;
     tracks = tracks_;
   }
 
-  // Deliver captured data to source provider, which stores the data into FIFO
-  // for WebAudio to fetch.
-  source_provider_->DeliverData(audio_source, audio_delay_milliseconds,
-                                current_volume, key_pressed);
-
   // Feed the data to the tracks.
   for (TrackList::const_iterator it = tracks.begin();
        it != tracks.end();
        ++it) {
     (*it)->Capture(audio_source, audio_delay_milliseconds,
                    current_volume, key_pressed);
   }
 }
 
 void WebRtcAudioCapturer::OnCaptureError() {
@@ skipping 27 matching lines @@
   // Use the native hardware buffer size in non peer connection mode on Mac.
   if (!peer_connection_mode_ && hardware_buffer_size_)
     return hardware_buffer_size_;
 #endif
 
   // WebRtc is running at a buffer size of 10ms data. Use a multiple of 10ms
   // as the buffer size to achieve the best performance for WebRtc.
   return (sample_rate / 100);
 }
 
+void WebRtcAudioCapturer::GetAudioProcessingParams(
+    int* delay_ms, int* volume, bool* key_pressed) {
+  base::AutoLock auto_lock(lock_);
+  *delay_ms = audio_delay_ms_;
+  *volume = volume_;
+  *key_pressed = key_pressed_;
+}
+
 }  // namespace content