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// Copyright 2014 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/modules/mediastream/media_stream_renderer_factory.h"
#include <utility>
#include "base/memory/scoped_refptr.h"
#include "base/task/sequenced_task_runner.h"
#include "base/task/single_thread_task_runner.h"
#include "third_party/blink/public/platform/modules/mediastream/web_media_stream.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/public/platform/platform.h"
#include "third_party/blink/public/web/web_local_frame.h"
#include "third_party/blink/renderer/core/execution_context/execution_context.h"
#include "third_party/blink/renderer/core/frame/local_dom_window.h"
#include "third_party/blink/renderer/core/frame/local_frame.h"
#include "third_party/blink/renderer/modules/mediastream/media_stream_video_renderer_sink.h"
#include "third_party/blink/renderer/modules/mediastream/media_stream_video_track.h"
#include "third_party/blink/renderer/modules/mediastream/track_audio_renderer.h"
#include "third_party/blink/renderer/modules/peerconnection/peer_connection_dependency_factory.h"
#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_device_impl.h"
#include "third_party/blink/renderer/modules/webrtc/webrtc_audio_renderer.h"
#include "third_party/blink/renderer/platform/mediastream/media_stream_audio_track.h"
#include "third_party/blink/renderer/platform/mediastream/media_stream_descriptor.h"
#include "third_party/blink/renderer/platform/webrtc/peer_connection_remote_audio_source.h"
#include "third_party/blink/renderer/platform/wtf/text/wtf_string.h"
#include "third_party/webrtc/api/media_stream_interface.h"
namespace blink {
namespace {
// Returns a valid session id if a single WebRTC capture device is currently
// open (and then the matching session_id), otherwise 0.
// This is used to pass on a session id to an audio renderer, so that audio will
// be rendered to a matching output device, should one exist.
// Note that if there are more than one open capture devices the function
// will not be able to pick an appropriate device and return 0.
base::UnguessableToken GetSessionIdForWebRtcAudioRenderer(
ExecutionContext& context) {
WebRtcAudioDeviceImpl* audio_device =
PeerConnectionDependencyFactory::From(context).GetWebRtcAudioDevice();
return audio_device
? audio_device->GetAuthorizedDeviceSessionIdForAudioRenderer()
: base::UnguessableToken();
}
void SendLogMessage(const WTF::String& message) {
WebRtcLogMessage("MSRF::" + message.Utf8());
}
} // namespace
MediaStreamRendererFactory::MediaStreamRendererFactory() {}
MediaStreamRendererFactory::~MediaStreamRendererFactory() {}
scoped_refptr<MediaStreamVideoRenderer>
MediaStreamRendererFactory::GetVideoRenderer(
const WebMediaStream& web_stream,
const MediaStreamVideoRenderer::RepaintCB& repaint_cb,
scoped_refptr<base::SequencedTaskRunner> video_task_runner,
scoped_refptr<base::SingleThreadTaskRunner> main_render_task_runner) {
DCHECK(!web_stream.IsNull());
DVLOG(1) << "MediaStreamRendererFactory::GetVideoRenderer stream:"
<< web_stream.Id().Utf8();
MediaStreamDescriptor& descriptor = *web_stream;
auto video_components = descriptor.VideoComponents();
if (video_components.empty() ||
!MediaStreamVideoTrack::GetTrack(
WebMediaStreamTrack(video_components[0].Get()))) {
return nullptr;
}
return base::MakeRefCounted<MediaStreamVideoRendererSink>(
video_components[0].Get(), repaint_cb, std::move(video_task_runner),
std::move(main_render_task_runner));
}
scoped_refptr<MediaStreamAudioRenderer>
MediaStreamRendererFactory::GetAudioRenderer(
const WebMediaStream& web_stream,
WebLocalFrame* web_frame,
const WebString& device_id,
base::RepeatingCallback<void()> on_render_error_callback) {
DCHECK(!web_stream.IsNull());
SendLogMessage(String::Format("%s({web_stream_id=%s}, {device_id=%s})",
__func__, web_stream.Id().Utf8().c_str(),
device_id.Utf8().c_str()));
MediaStreamDescriptor& descriptor = *web_stream;
auto audio_components = descriptor.AudioComponents();
if (audio_components.empty()) {
// The stream contains no audio tracks. Log error message if the stream
// contains no video tracks either. Without this extra check, video-only
// streams would generate error messages at this stage and we want to
// avoid that.
auto video_tracks = descriptor.VideoComponents();
if (video_tracks.empty()) {
SendLogMessage(String::Format(
"%s => (ERROR: no audio tracks in media stream)", __func__));
}
return nullptr;
}
// TODO(crbug.com/400764478): We need to fix the data flow so that
// it works the same way for all track implementations, local, remote or what
// have you.
// In this function, we should simply create a renderer object that receives
// and mixes audio from all the tracks that belong to the media stream.
// For now, we have separate renderers depending on if the first audio track
// in the stream is local or remote.
MediaStreamAudioTrack* audio_track =
MediaStreamAudioTrack::From(audio_components[0].Get());
if (!audio_track) {
// This can happen if the track was cloned.
// TODO(tommi, perkj): Fix cloning of tracks to handle extra data too.
SendLogMessage(String::Format(
"%s => (ERROR: no native track for WebMediaStreamTrack)", __func__));
return nullptr;
}
auto* frame = To<LocalFrame>(WebLocalFrame::ToCoreFrame(*web_frame));
DCHECK(frame);
// If the track has a local source, or is a remote track that does not use the
// WebRTC audio pipeline, return a new TrackAudioRenderer instance.
if (!PeerConnectionRemoteAudioTrack::From(audio_track)) {
// TODO(xians): Add support for the case where the media stream contains
// multiple audio tracks.
SendLogMessage(String::Format(
"%s => (creating TrackAudioRenderer for %s audio track)", __func__,
audio_track->is_local_track() ? "local" : "remote"));
return base::MakeRefCounted<TrackAudioRenderer>(
audio_components[0].Get(), *frame, String(device_id),
std::move(on_render_error_callback));
}
// Get the AudioDevice associated with the frame where this track was created,
// in case the track has been moved to eg a same origin iframe. Without this,
// one can get into a situation where media is piped to a different audio
// device to that where control signals are sent, leading to no audio being
// played out - see crbug/1239207.
WebLocalFrame* track_creation_frame =
audio_components[0].Get()->CreationFrame();
if (track_creation_frame) {
frame = To<LocalFrame>(WebLocalFrame::ToCoreFrame(*track_creation_frame));
}
// This is a remote WebRTC media stream.
WebRtcAudioDeviceImpl* audio_device =
PeerConnectionDependencyFactory::From(*frame->DomWindow())
.GetWebRtcAudioDevice();
DCHECK(audio_device);
SendLogMessage(String::Format(
"%s => (media stream is a remote WebRTC stream)", __func__));
// Share the existing renderer if any, otherwise create a new one.
scoped_refptr<WebRtcAudioRenderer> renderer(audio_device->renderer());
if (renderer) {
SendLogMessage(String::Format(
"%s => (using existing WebRtcAudioRenderer for remote stream)",
__func__));
} else {
SendLogMessage(String::Format(
"%s => (creating new WebRtcAudioRenderer for remote stream)",
__func__));
renderer = base::MakeRefCounted<WebRtcAudioRenderer>(
PeerConnectionDependencyFactory::From(*frame->DomWindow())
.GetWebRtcSignalingTaskRunner(),
web_stream, *web_frame,
GetSessionIdForWebRtcAudioRenderer(*frame->DomWindow()),
String(device_id), std::move(on_render_error_callback));
if (!audio_device->SetAudioRenderer(renderer.get())) {
SendLogMessage(String::Format(
"%s => (ERROR: WRADI::SetAudioRenderer failed)", __func__));
return nullptr;
}
}
auto ret = renderer->CreateSharedAudioRendererProxy(web_stream);
if (!ret) {
SendLogMessage(String::Format(
"%s => (ERROR: CreateSharedAudioRendererProxy failed)", __func__));
}
return ret;
}
} // namespace blink
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