MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc_audio_device_impl.cc

Index: content/renderer/media/webrtc_audio_device_impl.cc
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index b5dc5008ad7ff831a8de3c05ff9d9795c005d950..018716d30651124d523c561086bd20b80b3fc4f1 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -9,7 +9,7 @@
 #include "base/strings/string_util.h"
 #include "base/win/windows_version.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/webrtc/processed_local_audio_source.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_parameters.h"
@@ -170,15 +170,9 @@ int32_t WebRtcAudioDeviceImpl::Terminate() {
   DCHECK(!renderer_.get() || !renderer_->IsStarted())
       << "The shared audio renderer shouldn't be running";
 
-  // Stop all the capturers to ensure no further OnData() and
-  // RemoveAudioCapturer() callback.
-  // Cache the capturers in a local list since WebRtcAudioCapturer::Stop()
-  // will trigger RemoveAudioCapturer() callback.
-  CapturerList capturers;
-  capturers.swap(capturers_);
-  for (CapturerList::const_iterator iter = capturers.begin();
-       iter != capturers.end(); ++iter) {
-    (*iter)->Stop();
+  {
+    base::AutoLock auto_lock(lock_);
+    capturers_.clear();
   }
 
   initialized_ = false;
@@ -294,11 +288,10 @@ int32_t WebRtcAudioDeviceImpl::SetMicrophoneVolume(uint32_t volume) {
 
   // Only one microphone is supported at the moment, which is represented by
   // the default capturer.
-  scoped_refptr<WebRtcAudioCapturer> capturer(GetDefaultCapturer());
-  if (!capturer.get())
+  base::AutoLock auto_lock(lock_);
+  if (capturers_.empty())
     return -1;
-
-  capturer->SetVolume(volume);
+  capturers_.back()->SetVolume(volume);
   return 0;
 }
 
@@ -309,12 +302,10 @@ int32_t WebRtcAudioDeviceImpl::MicrophoneVolume(uint32_t* volume) const {
   // We only support one microphone now, which is accessed via the default
   // capturer.
   DCHECK(initialized_);
-  scoped_refptr<WebRtcAudioCapturer> capturer(GetDefaultCapturer());
-  if (!capturer.get())
+  base::AutoLock auto_lock(lock_);
+  if (capturers_.empty())
     return -1;
-
-  *volume = static_cast<uint32_t>(capturer->Volume());
-
+  *volume = static_cast<uint32_t>(capturers_.back()->Volume());
   return 0;
 }
 
@@ -352,11 +343,10 @@ int32_t WebRtcAudioDeviceImpl::StereoRecordingIsAvailable(
 
   // TODO(xians): These kind of hardware methods do not make much sense since we
   // support multiple sources. Remove or figure out new APIs for such methods.
-  scoped_refptr<WebRtcAudioCapturer> capturer(GetDefaultCapturer());
-  if (!capturer.get())
+  base::AutoLock auto_lock(lock_);
+  if (capturers_.empty())
     return -1;
-
-  *available = (capturer->source_audio_parameters().channels() == 2);
+  *available = (capturers_.back()->GetInputAudioParameters().channels() == 2);
   return 0;
 }
 
@@ -371,7 +361,7 @@ int32_t WebRtcAudioDeviceImpl::RecordingDelay(uint16_t* delay_ms) const {
   DCHECK(signaling_thread_checker_.CalledOnValidThread());
 
   // There is no way to report a correct delay value to WebRTC since there
-  // might be multiple WebRtcAudioCapturer instances.
+  // might be multiple ProcessedLocalAudioSource instances.
   NOTREACHED();
   return -1;
 }
@@ -380,12 +370,14 @@ int32_t WebRtcAudioDeviceImpl::RecordingSampleRate(
     uint32_t* sample_rate) const {
   DCHECK(signaling_thread_checker_.CalledOnValidThread());
   // We use the default capturer as the recording sample rate.
-  scoped_refptr<WebRtcAudioCapturer> capturer(GetDefaultCapturer());
-  if (!capturer.get())
+  base::AutoLock auto_lock(lock_);
+  if (capturers_.empty())
     return -1;
-
-  *sample_rate = static_cast<uint32_t>(
-      capturer->source_audio_parameters().sample_rate());
+  const media::AudioParameters& params =
+      capturers_.back()->GetInputAudioParameters();
+  if (!params.IsValid())
+    return -1;
+  *sample_rate = static_cast<uint32_t>(params.sample_rate());
   return 0;
 }
 
@@ -434,11 +426,10 @@ bool WebRtcAudioDeviceImpl::SetAudioRenderer(WebRtcAudioRenderer* renderer) {
 }
 
 void WebRtcAudioDeviceImpl::AddAudioCapturer(
-    const scoped_refptr<WebRtcAudioCapturer>& capturer) {
+    ProcessedLocalAudioSource* capturer) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioDeviceImpl::AddAudioCapturer()";
-  DCHECK(capturer.get());
-  DCHECK(!capturer->device_id().empty());
+  DCHECK(capturer);
 
   base::AutoLock auto_lock(lock_);
   DCHECK(std::find(capturers_.begin(), capturers_.end(), capturer) ==
@@ -447,28 +438,14 @@ void WebRtcAudioDeviceImpl::AddAudioCapturer(
 }
 
 void WebRtcAudioDeviceImpl::RemoveAudioCapturer(
-    const scoped_refptr<WebRtcAudioCapturer>& capturer) {
+    ProcessedLocalAudioSource* capturer) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   DVLOG(1) << "WebRtcAudioDeviceImpl::AddAudioCapturer()";
-  DCHECK(capturer.get());
+  DCHECK(capturer);
   base::AutoLock auto_lock(lock_);
   capturers_.remove(capturer);
 }
 
-scoped_refptr<WebRtcAudioCapturer>
-WebRtcAudioDeviceImpl::GetDefaultCapturer() const {
-  // Called on the signaling thread (during initialization), worker
-  // thread during capture or main thread for a WebAudio source.
-  // We can't DCHECK on those three checks here since GetDefaultCapturer
-  // may be the first call and therefore could incorrectly initialize the
-  // thread checkers.
-  DCHECK(initialized_);
-  base::AutoLock auto_lock(lock_);
-  // Use the last |capturer| which is from the latest getUserMedia call as
-  // the default capture device.
-  return capturers_.empty() ? NULL : capturers_.back();
-}
-
 void WebRtcAudioDeviceImpl::AddPlayoutSink(
     WebRtcPlayoutDataSource::Sink* sink) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
@@ -498,8 +475,19 @@ bool WebRtcAudioDeviceImpl::GetAuthorizedDeviceInfoForAudioRenderer(
   if (capturers_.size() != 1)
     return false;
 
-  return capturers_.back()->GetPairedOutputParameters(
-      session_id, output_sample_rate, output_frames_per_buffer);
+  // Don't set output parameters unless all of them are valid.
+  const StreamDeviceInfo& device_info = capturers_.back()->device_info();
+  if (device_info.session_id <= 0 ||
+      !device_info.device.matched_output.sample_rate ||
+      !device_info.device.matched_output.frames_per_buffer)
+    return false;
+
+  *session_id = device_info.session_id;
+  *output_sample_rate = device_info.device.matched_output.sample_rate;
+  *output_frames_per_buffer =
+      device_info.device.matched_output.frames_per_buffer;
+
+  return true;
 }
 
 }  // namespace content