Adding JSEP PeerConnection glue - attempt 2

content/renderer/media/peer_connection_handler_base.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/peer_connection_handler_base.h"
+
+#include "base/logging.h"
+#include "base/utf_string_conversions.h"
+#include "content/renderer/media/media_stream_dependency_factory.h"
+#include "content/renderer/media/media_stream_impl.h"
+#include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamDescriptor.h"
+#include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebMediaStreamSource.h"
+#include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebString.h"
+
+PeerConnectionHandlerBase::PeerConnectionHandlerBase(
+    MediaStreamImpl* msi,
+    MediaStreamDependencyFactory* dependency_factory)
+    : media_stream_impl_(msi),
+      dependency_factory_(dependency_factory),
+      message_loop_proxy_(base::MessageLoopProxy::current()) {
+}
+
+PeerConnectionHandlerBase::~PeerConnectionHandlerBase() {
+}
+
+bool PeerConnectionHandlerBase::HasStream(const std::string& stream_label) {
+  webrtc::MediaStreamInterface* stream =
+      native_peer_connection_->remote_streams()->find(stream_label);
+  return stream != NULL;
+}
+
+void PeerConnectionHandlerBase::SetVideoRenderer(
+    const std::string& stream_label,
+    webrtc::VideoRendererWrapperInterface* renderer) {
+  webrtc::MediaStreamInterface* stream =
+      native_peer_connection_->remote_streams()->find(stream_label);
+  webrtc::VideoTracks* video_tracks = stream->video_tracks();
+  // We assume there is only one enabled video track.
+  for (size_t i = 0; i < video_tracks->count(); ++i) {
+    webrtc::VideoTrackInterface* video_track = video_tracks->at(i);
+    if (video_track->enabled()) {
+      video_track->SetRenderer(renderer);
+      return;
+    }
+  }
+  DVLOG(1) << "No enabled video track.";
+}
+
+void PeerConnectionHandlerBase::AddStream(
+    const WebKit::WebMediaStreamDescriptor& stream) {
+  talk_base::scoped_refptr<webrtc::LocalMediaStreamInterface> native_stream =
+      dependency_factory_->CreateLocalMediaStream(UTF16ToUTF8(stream.label()));
+  WebKit::WebVector<WebKit::WebMediaStreamSource> source_vector;
+  stream.sources(source_vector);
+
+  // Get and add all tracks.
+  for (size_t i = 0; i < source_vector.size(); ++i) {
+    webrtc::MediaStreamTrackInterface* track = media_stream_impl_
+        ->GetLocalMediaStreamTrack(UTF16ToUTF8(source_vector[i].id()));
+    DCHECK(track);
+    if (source_vector[i].type() == WebKit::WebMediaStreamSource::TypeVideo) {
+      native_stream->AddTrack(static_cast<webrtc::VideoTrackInterface*>(track));
+    } else {
+      DCHECK(source_vector[i].type() ==
+          WebKit::WebMediaStreamSource::TypeAudio);
+      native_stream->AddTrack(static_cast<webrtc::AudioTrackInterface*>(track));
+    }
+  }
+
+  native_peer_connection_->AddStream(native_stream);
+}
+
+void PeerConnectionHandlerBase::RemoveStream(
+    const WebKit::WebMediaStreamDescriptor& stream) {
+  talk_base::scoped_refptr<webrtc::StreamCollectionInterface> native_streams =
+      native_peer_connection_->local_streams();
+  if (!native_streams)
+    return;
+  // TODO(perkj): Change libJingle PeerConnection::RemoveStream API to take a
+  // label as input instead of stream and return bool.
+  webrtc::LocalMediaStreamInterface* native_stream =
+      static_cast<webrtc::LocalMediaStreamInterface*>(native_streams->find(
+          UTF16ToUTF8(stream.label())));
+  DCHECK(native_stream);
+  native_peer_connection_->RemoveStream(native_stream);
+}
+
+WebKit::WebMediaStreamDescriptor
+PeerConnectionHandlerBase::CreateWebKitStreamDescriptor(
+    webrtc::MediaStreamInterface* stream) {
+  webrtc::AudioTracks* audio_tracks = stream->audio_tracks();
+  webrtc::VideoTracks* video_tracks = stream->video_tracks();
+  WebKit::WebVector<WebKit::WebMediaStreamSource> source_vector(
+      audio_tracks->count() + video_tracks->count());
+
+  // Add audio tracks.
+  size_t i = 0;
+  for (; i < audio_tracks->count(); ++i) {
+    webrtc::AudioTrackInterface* audio_track = audio_tracks->at(i);
+    DCHECK(audio_track);
+    source_vector[i].initialize(
+        // TODO(grunell): Set id to something unique.
+        UTF8ToUTF16(audio_track->label()),
+        WebKit::WebMediaStreamSource::TypeAudio,
+        UTF8ToUTF16(audio_track->label()));
+  }
+
+  // Add video tracks.
+  for (i = 0; i < video_tracks->count(); ++i) {
+    webrtc::VideoTrackInterface* video_track = video_tracks->at(i);
+    DCHECK(video_track);
+    source_vector[audio_tracks->count() + i].initialize(
+        // TODO(grunell): Set id to something unique.
+        UTF8ToUTF16(video_track->label()),
+        WebKit::WebMediaStreamSource::TypeVideo,
+        UTF8ToUTF16(video_track->label()));
+  }
+
+  WebKit::WebMediaStreamDescriptor descriptor;
+  descriptor.initialize(UTF8ToUTF16(stream->label()), source_vector);
+
+  return descriptor;
+}