Implicit audio output device selection for getUserMedia.

content/renderer/media/media_stream_impl.cc

Index: content/renderer/media/media_stream_impl.cc
diff --git a/content/renderer/media/media_stream_impl.cc b/content/renderer/media/media_stream_impl.cc
index 9eab908f5f90e64927df8e065b0a605a71145c31..f1504cf2c9bbf84c767805c672df6e89f70659e8 100644
--- a/content/renderer/media/media_stream_impl.cc
+++ b/content/renderer/media/media_stream_impl.cc
@@ -21,6 +21,8 @@
 #include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/renderer/media/webrtc_local_audio_renderer.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
+#include "content/renderer/render_thread_impl.h"
+#include "media/base/audio_hardware_config.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
@@ -592,7 +594,22 @@ scoped_refptr<WebRtcAudioRenderer> MediaStreamImpl::CreateRemoteAudioRenderer(
   DVLOG(1) << "MediaStreamImpl::CreateRemoteAudioRenderer label:"
            << stream->label();
 
-  return new WebRtcAudioRenderer(RenderViewObserver::routing_id());
+  int session_id = -1, sample_rate = 0, buffer_size = 0;
+  if (!GetAuthorizedInputDeviceSessionIdForAudioRenderer(&session_id,
+                                                         &sample_rate,
+                                                         &buffer_size)) {
+    // Fetch the default audio output hardware config.
+    media::AudioHardwareConfig* hardware_config =
+        RenderThreadImpl::current()->GetAudioHardwareConfig();
+    sample_rate = hardware_config->GetOutputSampleRate();
+    buffer_size = hardware_config->GetOutputBufferSize();
+    // Use session_id of 0 to indicate no association with the capture device

[Jói, 2013/09/06 14:49:26]

Above you use a guard of -1, which is correct?

[tommi (sloooow) - chröme, 2013/09/06 16:56:54]

I changed it to 0 for consistency with what's used in AudioOutputDevice:
https://code.google.com/p/chromium/codesearch#chromium/src/media/audio/audio_...

+    // (i.e. use the default device).
+    session_id = 0;
+  }
+
+  return new WebRtcAudioRenderer(RenderViewObserver::routing_id(),
+      session_id, sample_rate, buffer_size);
 }
 
 scoped_refptr<WebRtcLocalAudioRenderer>
@@ -611,11 +628,28 @@ MediaStreamImpl::CreateLocalAudioRenderer(
            << "audio_track.id     : " << audio_track->id()
            << "audio_track.enabled: " << audio_track->enabled();
 
+  int session_id = -1, sample_rate = 0, buffer_size = 0;

[Jói, 2013/09/06 14:49:26]

Same code, DRY.

[tommi (sloooow) - chröme, 2013/09/06 16:56:54]

Done.

+  if (!GetAuthorizedInputDeviceSessionIdForAudioRenderer(&session_id,
+                                                         &sample_rate,
+                                                         &buffer_size)) {
+    // Fetch the default audio output hardware config.
+    media::AudioHardwareConfig* hardware_config =
+        RenderThreadImpl::current()->GetAudioHardwareConfig();
+    sample_rate = hardware_config->GetOutputSampleRate();
+    buffer_size = hardware_config->GetOutputBufferSize();
+    // Use session_id of 0 to indicate no association with the capture device
+    // (i.e. use the default device).
+    session_id = 0;
+  }
+
   // Create a new WebRtcLocalAudioRenderer instance and connect it to the
   // existing WebRtcAudioCapturer so that the renderer can use it as source.
   return new WebRtcLocalAudioRenderer(
       static_cast<WebRtcLocalAudioTrack*>(audio_track),
-      RenderViewObserver::routing_id());
+      RenderViewObserver::routing_id(),
+      session_id,
+      sample_rate,
+      buffer_size);
 }
 
 void MediaStreamImpl::StopLocalAudioTrack(
@@ -639,6 +673,47 @@ void MediaStreamImpl::StopLocalAudioTrack(
   }
 }
 
+bool MediaStreamImpl::GetAuthorizedInputDeviceSessionIdForAudioRenderer(

[no longer working on chromium, 2013/09/06 15:17:37]

The name does not completely fit to its functionalities, you are getting the
device related info like session_id, sample_rate, and buffer size. I will
suggest renaming it to GetAurhorizedDeviceInfoForAudioRenderer

[tommi (sloooow) - chröme, 2013/09/06 16:56:54]

Done (fixed typo).

+    int* session_id,
+    int* output_sample_rate,
+    int* output_frames_per_buffer) {
+  DCHECK(CalledOnValidThread());
+
+  const StreamDeviceInfo* device_info = NULL;
+  WebKit::WebString device_id;
+  UserMediaRequests::iterator it = user_media_requests_.begin();
+  for (; it != user_media_requests_.end(); ++it) {
+    UserMediaRequestInfo* request = (*it);
+    for (size_t i = 0; i < request->audio_sources.size(); ++i) {
+      const WebKit::WebMediaStreamSource& source = request->audio_sources[i];
+      if (source.readyState() == WebKit::WebMediaStreamSource::ReadyStateEnded)
+        continue;
+
+      if (!device_id.isEmpty() && !device_id.equals(source.deviceId())) {
+        DLOG(WARNING)

[Jói, 2013/09/06 14:49:26]

Not sure this is "bad" enough to demand a warning; maybe INFO?

[tommi (sloooow) - chröme, 2013/09/06 16:56:54]

Yeah, that's why I picked DLOG so that while debugging we would notice it
(likely only webrtc engs would run into this), but changed it to DVLOG(1).

+            << "Multiple capture devices are open so we can't pick a "
+            "session for a matching output device.";
+        return false;
+      }
+
+      device_id = source.deviceId();

[no longer working on chromium, 2013/09/06 15:17:37]

this should be the session_id

[tommi (sloooow) - chröme, 2013/09/06 16:56:54]

Renamed the variable.  Also added a todo to store the session_id in the
MediaStreamSourceExtraData struct.

+      content::MediaStreamSourceExtraData* extra_data =
+          static_cast<content::MediaStreamSourceExtraData*>(source.extraData());
+      device_info = &extra_data->device_info();
+    }
+  }
+
+  if (device_id.isEmpty() || !device_info)
+    return false;
+
+  base::StringToInt(UTF16ToUTF8(device_id), 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;
+}
+
 MediaStreamSourceExtraData::MediaStreamSourceExtraData(
     const StreamDeviceInfo& device_info,
     const WebKit::WebMediaStreamSource& webkit_source)