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 routing_id,
  62     int render_frame_id) {
  63   if (stream->GetAudioTracks().empty())
  64     return NULL;
  65
  66   DVLOG(1) << "MediaStreamRendererFactory::CreateRemoteAudioRenderer label:"
  67            << stream->label();
  68
  69   // TODO(tommi): Change the default value of session_id to be
  70   // StreamDeviceInfo::kNoId.  Also update AudioOutputDevice etc.
  71   int session_id = 0, sample_rate = 0, buffer_size = 0;
  72   if (!GetAuthorizedDeviceInfoForAudioRenderer(&session_id,
  73                                                &sample_rate,
  74                                                &buffer_size)) {
  75     GetDefaultOutputDeviceParams(&sample_rate, &buffer_size);
  76   }
  77
  78   return new WebRtcAudioRenderer(
  79       stream, routing_id, render_frame_id,  session_id,
  80       sample_rate, buffer_size);
  81 }
  82
  83
  84 scoped_refptr<WebRtcLocalAudioRenderer> CreateLocalAudioRenderer(
  85     const blink::WebMediaStreamTrack& audio_track,
  86     int routing_id,
  87     int render_frame_id) {
  88   DVLOG(1) << "MediaStreamRendererFactory::CreateLocalAudioRenderer";
  89
  90   int session_id = 0, sample_rate = 0, buffer_size = 0;
  91   if (!GetAuthorizedDeviceInfoForAudioRenderer(&session_id,
  92                                                &sample_rate,
  93                                                &buffer_size)) {
  94     GetDefaultOutputDeviceParams(&sample_rate, &buffer_size);
  95   }
  96
  97   // Create a new WebRtcLocalAudioRenderer instance and connect it to the
  98   // existing WebRtcAudioCapturer so that the renderer can use it as source.
  99   return new WebRtcLocalAudioRenderer(
 100       audio_track,
 101       routing_id,
 102       render_frame_id,
 103       session_id,
 104       buffer_size);
 105 }
 106
 107 }  // namespace
 108
 109
 110 MediaStreamRendererFactory::MediaStreamRendererFactory() {
 111 }
 112
 113 MediaStreamRendererFactory::~MediaStreamRendererFactory() {
 114 }
 115
 116 scoped_refptr<VideoFrameProvider>
 117 MediaStreamRendererFactory::GetVideoFrameProvider(
 118     const GURL& url,
 119     const base::Closure& error_cb,
 120     const VideoFrameProvider::RepaintCB& repaint_cb) {
 121   blink::WebMediaStream web_stream =
 122       blink::WebMediaStreamRegistry::lookupMediaStreamDescriptor(url);
 123   DCHECK(!web_stream.isNull());
 124
 125   DVLOG(1) << "MediaStreamRendererFactory::GetVideoFrameProvider stream:"
 126            << base::UTF16ToUTF8(web_stream.id());
 127
 128   blink::WebVector<blink::WebMediaStreamTrack> video_tracks;
 129   web_stream.videoTracks(video_tracks);
 130   if (video_tracks.isEmpty() ||
 131       !MediaStreamVideoTrack::GetTrack(video_tracks[0])) {
 132     return NULL;
 133   }
 134
 135   return new RTCVideoRenderer(video_tracks[0], error_cb, repaint_cb);
 136 }
 137
 138 scoped_refptr<MediaStreamAudioRenderer>
 139 MediaStreamRendererFactory::GetAudioRenderer(
 140     const GURL& url, int render_view_id, int render_frame_id) {
 141   blink::WebMediaStream web_stream =
 142       blink::WebMediaStreamRegistry::lookupMediaStreamDescriptor(url);
 143
 144   if (web_stream.isNull() || !web_stream.extraData())
 145     return NULL;  // This is not a valid stream.
 146
 147   DVLOG(1) << "MediaStreamRendererFactory::GetAudioRenderer stream:"
 148            << base::UTF16ToUTF8(web_stream.id());
 149
 150   MediaStream* native_stream = MediaStream::GetMediaStream(web_stream);
 151
 152   // TODO(tommi): MediaStreams do not have a 'local or not' concept.
 153   // Tracks _might_, but even so, we need to fix the data flow so that
 154   // it works the same way for all track implementations, local, remote or what
 155   // have you.
 156   // In this function, we should simply create a renderer object that receives
 157   // and mixes audio from all the tracks that belong to the media stream.
 158   // We need to remove the |is_local| property from MediaStreamExtraData since
 159   // this concept is peerconnection specific (is a previously recorded stream
 160   // local or remote?).
 161   if (native_stream->is_local()) {
 162     // Create the local audio renderer if the stream contains audio tracks.
 163     blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
 164     web_stream.audioTracks(audio_tracks);
 165     if (audio_tracks.isEmpty())
 166       return NULL;
 167
 168     // TODO(xians): Add support for the case where the media stream contains
 169     // multiple audio tracks.
 170     return CreateLocalAudioRenderer(audio_tracks[0], render_view_id,
 171                                     render_frame_id);
 172   }
 173
 174   webrtc::MediaStreamInterface* stream =
 175       MediaStream::GetAdapter(web_stream);
 176   if (stream->GetAudioTracks().empty())
 177     return NULL;
 178
 179   // This is a remote WebRTC media stream.
 180   WebRtcAudioDeviceImpl* audio_device =
 181       GetPeerConnectionDependencyFactory()->GetWebRtcAudioDevice();
 182
 183   // Share the existing renderer if any, otherwise create a new one.
 184   scoped_refptr<WebRtcAudioRenderer> renderer(audio_device->renderer());
 185   if (!renderer.get()) {
 186     renderer = CreateRemoteAudioRenderer(stream, render_view_id,
 187                                          render_frame_id);
 188
 189     if (renderer.get() && !audio_device->SetAudioRenderer(renderer.get()))
 190       renderer = NULL;
 191   }
 192
 193   return renderer.get() ?
 194       renderer->CreateSharedAudioRendererProxy(stream) : NULL;
 195 }
 196
 197 }  // namespace content