content/renderer/media/media_stream_renderer_factory.cc

   1 // Copyright 2014 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "content/renderer/media/media_stream_renderer_factory.h"
   6
   7 #include "base/strings/utf_string_conversions.h"
   8 #include "content/renderer/media/media_stream.h"
   9 #include "content/renderer/media/media_stream_video_track.h"
  10 #include "content/renderer/media/rtc_video_renderer.h"
  11 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
  12 #include "content/renderer/media/webrtc_audio_renderer.h"
  13 #include "content/renderer/media/webrtc_local_audio_renderer.h"
  14 #include "content/renderer/render_thread_impl.h"
  15 #include "media/base/audio_hardware_config.h"
  16 #include "third_party/WebKit/public/platform/WebMediaStream.h"
  17 #include "third_party/WebKit/public/platform/WebURL.h"
  18 #include "third_party/WebKit/public/web/WebMediaStreamRegistry.h"
  19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  20
  21 namespace content {
  22
  23 namespace {
  24
  25 PeerConnectionDependencyFactory* GetPeerConnectionDependencyFactory() {
  26   return RenderThreadImpl::current()->GetPeerConnectionDependencyFactory();
  27 }
  28
  29 void GetDefaultOutputDeviceParams(
  30     int* output_sample_rate, int* output_buffer_size) {
  31   // Fetch the default audio output hardware config.
  32   media::AudioHardwareConfig* hardware_config =
  33       RenderThreadImpl::current()->GetAudioHardwareConfig();
  34   *output_sample_rate = hardware_config->GetOutputSampleRate();
  35   *output_buffer_size = hardware_config->GetOutputBufferSize();
  36 }
  37
  38
  39 // Returns a valid session id if a single capture device is currently open
  40 // (and then the matching session_id), otherwise -1.
  41 // This is used to pass on a session id to a webrtc audio renderer (either
  42 // local or remote), so that audio will be rendered to a matching output
  43 // device, should one exist.
  44 // Note that if there are more than one open capture devices the function
  45 // will not be able to pick an appropriate device and return false.
  46 bool GetAuthorizedDeviceInfoForAudioRenderer(
  47     int* session_id,
  48     int* output_sample_rate,
  49     int* output_frames_per_buffer) {
  50   WebRtcAudioDeviceImpl* audio_device =
  51       GetPeerConnectionDependencyFactory()->GetWebRtcAudioDevice();
  52   if (!audio_device)
  53     return false;
  54
  55   return audio_device->GetAuthorizedDeviceInfoForAudioRenderer(
  56       session_id, output_sample_rate, output_frames_per_buffer);
  57 }
  58
  59 scoped_refptr<WebRtcAudioRenderer> CreateRemoteAudioRenderer(
  60     webrtc::MediaStreamInterface* stream,
  61     int render_frame_id) {
  62   if (stream->GetAudioTracks().empty())
  63     return NULL;
  64
  65   DVLOG(1) << "MediaStreamRendererFactory::CreateRemoteAudioRenderer label:"
  66            << stream->label();
  67
  68   // TODO(tommi): Change the default value of session_id to be
  69   // StreamDeviceInfo::kNoId.  Also update AudioOutputDevice etc.
  70   int session_id = 0, sample_rate = 0, buffer_size = 0;
  71   if (!GetAuthorizedDeviceInfoForAudioRenderer(&session_id,
  72                                                &sample_rate,
  73                                                &buffer_size)) {
  74     GetDefaultOutputDeviceParams(&sample_rate, &buffer_size);
  75   }
  76
  77   return new WebRtcAudioRenderer(
  78       GetPeerConnectionDependencyFactory()->GetWebRtcSignalingThread(), stream,
  79       render_frame_id, session_id, sample_rate, buffer_size);
  80 }
  81
  82
  83 scoped_refptr<WebRtcLocalAudioRenderer> CreateLocalAudioRenderer(
  84     const blink::WebMediaStreamTrack& audio_track,
  85     int render_frame_id) {
  86   DVLOG(1) << "MediaStreamRendererFactory::CreateLocalAudioRenderer";
  87
  88   int session_id = 0, sample_rate = 0, buffer_size = 0;
  89   if (!GetAuthorizedDeviceInfoForAudioRenderer(&session_id,
  90                                                &sample_rate,
  91                                                &buffer_size)) {
  92     GetDefaultOutputDeviceParams(&sample_rate, &buffer_size);
  93   }
  94
  95   // Create a new WebRtcLocalAudioRenderer instance and connect it to the
  96   // existing WebRtcAudioCapturer so that the renderer can use it as source.
  97   return new WebRtcLocalAudioRenderer(
  98       audio_track,
  99       render_frame_id,
 100       session_id,
 101       buffer_size);
 102 }
 103
 104 }  // namespace
 105
 106
 107 MediaStreamRendererFactory::MediaStreamRendererFactory() {
 108 }
 109
 110 MediaStreamRendererFactory::~MediaStreamRendererFactory() {
 111 }
 112
 113 scoped_refptr<VideoFrameProvider>
 114 MediaStreamRendererFactory::GetVideoFrameProvider(
 115     const GURL& url,
 116     const base::Closure& error_cb,
 117     const VideoFrameProvider::RepaintCB& repaint_cb) {
 118   blink::WebMediaStream web_stream =
 119       blink::WebMediaStreamRegistry::lookupMediaStreamDescriptor(url);
 120   DCHECK(!web_stream.isNull());
 121
 122   DVLOG(1) << "MediaStreamRendererFactory::GetVideoFrameProvider stream:"
 123            << base::UTF16ToUTF8(web_stream.id());
 124
 125   blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
 126   web_stream.videoTracks(video_tracks);
 127   if (video_tracks.isEmpty() ||
 128       !MediaStreamVideoTrack::GetTrack(video_tracks[0])) {
 129     return NULL;
 130   }
 131
 132   return new RTCVideoRenderer(video_tracks[0], error_cb, repaint_cb);
 133 }
 134
 135 scoped_refptr<MediaStreamAudioRenderer>
 136 MediaStreamRendererFactory::GetAudioRenderer(const GURL& url,
 137                                              int render_frame_id) {
 138   blink::WebMediaStream web_stream =
 139       blink::WebMediaStreamRegistry::lookupMediaStreamDescriptor(url);
 140
 141   if (web_stream.isNull() || !web_stream.extraData())
 142     return NULL;  // This is not a valid stream.
 143
 144   DVLOG(1) << "MediaStreamRendererFactory::GetAudioRenderer stream:"
 145            << base::UTF16ToUTF8(web_stream.id());
 146
 147   MediaStream* native_stream = MediaStream::GetMediaStream(web_stream);
 148
 149   // TODO(tommi): MediaStreams do not have a 'local or not' concept.
 150   // Tracks _might_, but even so, we need to fix the data flow so that
 151   // it works the same way for all track implementations, local, remote or what
 152   // have you.
 153   // In this function, we should simply create a renderer object that receives
 154   // and mixes audio from all the tracks that belong to the media stream.
 155   // We need to remove the |is_local| property from MediaStreamExtraData since
 156   // this concept is peerconnection specific (is a previously recorded stream
 157   // local or remote?).
 158   if (native_stream->is_local()) {
 159     // Create the local audio renderer if the stream contains audio tracks.
 160     blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
 161     web_stream.audioTracks(audio_tracks);
 162     if (audio_tracks.isEmpty())
 163       return NULL;
 164
 165     // TODO(xians): Add support for the case where the media stream contains
 166     // multiple audio tracks.
 167     return CreateLocalAudioRenderer(audio_tracks[0], render_frame_id);
 168   }
 169
 170   webrtc::MediaStreamInterface* stream =
 171       MediaStream::GetAdapter(web_stream);
 172   if (stream->GetAudioTracks().empty())
 173     return NULL;
 174
 175   // This is a remote WebRTC media stream.
 176   WebRtcAudioDeviceImpl* audio_device =
 177       GetPeerConnectionDependencyFactory()->GetWebRtcAudioDevice();
 178
 179   // Share the existing renderer if any, otherwise create a new one.
 180   scoped_refptr<WebRtcAudioRenderer> renderer(audio_device->renderer());
 181   if (!renderer.get()) {
 182     renderer = CreateRemoteAudioRenderer(stream, render_frame_id);
 183
 184     if (renderer.get() && !audio_device->SetAudioRenderer(renderer.get()))
 185       renderer = NULL;
 186   }
 187
 188   return renderer.get() ?
 189       renderer->CreateSharedAudioRendererProxy(stream) : NULL;
 190 }
 191
 192 }  // namespace content