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-rw-r--r--chrome/renderer/extensions/cast_streaming_native_handler.cc47
-rw-r--r--chrome/renderer/extensions/cast_streaming_native_handler.h4
-rw-r--r--chrome/renderer/extensions/chrome_v8_context.cc12
-rw-r--r--chrome/renderer/extensions/chrome_v8_context.h2
-rw-r--r--chrome/renderer/media/cast_rtp_stream.cc60
-rw-r--r--chrome/renderer/media/cast_rtp_stream.h17
-rw-r--r--chrome/test/data/extensions/api_test/cast_streaming/basics.js52
7 files changed, 175 insertions, 19 deletions
diff --git a/chrome/renderer/extensions/cast_streaming_native_handler.cc b/chrome/renderer/extensions/cast_streaming_native_handler.cc
index 8170f63..71fd3e9 100644
--- a/chrome/renderer/extensions/cast_streaming_native_handler.cc
+++ b/chrome/renderer/extensions/cast_streaming_native_handler.cc
@@ -236,6 +236,38 @@ void CastStreamingNativeHandler::CallCreateCallback(
create_callback_.reset();
}
+void CastStreamingNativeHandler::CallStartCallback(int stream_id) {
+ v8::Isolate* isolate = context()->isolate();
+ v8::HandleScope handle_scope(isolate);
+ v8::Context::Scope context_scope(context()->v8_context());
+ v8::Handle<v8::Array> event_args = v8::Array::New(isolate, 1);
+ event_args->Set(0, v8::Integer::New(isolate, stream_id));
+ context()->DispatchEvent("cast.streaming.rtpStream.onStarted", event_args);
+}
+
+void CastStreamingNativeHandler::CallStopCallback(int stream_id) {
+ v8::Isolate* isolate = context()->isolate();
+ v8::HandleScope handle_scope(isolate);
+ v8::Context::Scope context_scope(context()->v8_context());
+ v8::Handle<v8::Array> event_args = v8::Array::New(isolate, 1);
+ event_args->Set(0, v8::Integer::New(isolate, stream_id));
+ context()->DispatchEvent("cast.streaming.rtpStream.onStopped", event_args);
+}
+
+void CastStreamingNativeHandler::CallErrorCallback(int stream_id,
+ const std::string& message) {
+ v8::Isolate* isolate = context()->isolate();
+ v8::HandleScope handle_scope(isolate);
+ v8::Context::Scope context_scope(context()->v8_context());
+ v8::Handle<v8::Array> event_args = v8::Array::New(isolate, 2);
+ event_args->Set(0, v8::Integer::New(isolate, stream_id));
+ event_args->Set(
+ 1,
+ v8::String::NewFromUtf8(
+ isolate, message.data(), v8::String::kNormalString, message.size()));
+ context()->DispatchEvent("cast.streaming.rtpStream.onError", event_args);
+}
+
void CastStreamingNativeHandler::DestroyCastRtpStream(
const v8::FunctionCallbackInfo<v8::Value>& args) {
CHECK_EQ(1, args.Length());
@@ -303,7 +335,20 @@ void CastStreamingNativeHandler::StartCastRtpStream(
v8::Isolate* isolate = context()->v8_context()->GetIsolate();
if (!ToCastRtpParamsOrThrow(isolate, *params, &cast_params))
return;
- transport->Start(cast_params);
+
+ base::Closure start_callback =
+ base::Bind(&CastStreamingNativeHandler::CallStartCallback,
+ weak_factory_.GetWeakPtr(),
+ transport_id);
+ base::Closure stop_callback =
+ base::Bind(&CastStreamingNativeHandler::CallStopCallback,
+ weak_factory_.GetWeakPtr(),
+ transport_id);
+ CastRtpStream::ErrorCallback error_callback =
+ base::Bind(&CastStreamingNativeHandler::CallErrorCallback,
+ weak_factory_.GetWeakPtr(),
+ transport_id);
+ transport->Start(cast_params, start_callback, stop_callback, error_callback);
}
void CastStreamingNativeHandler::StopCastRtpStream(
diff --git a/chrome/renderer/extensions/cast_streaming_native_handler.h b/chrome/renderer/extensions/cast_streaming_native_handler.h
index 02d8efa..9f41a92 100644
--- a/chrome/renderer/extensions/cast_streaming_native_handler.h
+++ b/chrome/renderer/extensions/cast_streaming_native_handler.h
@@ -54,6 +54,10 @@ class CastStreamingNativeHandler : public ObjectBackedNativeHandler {
scoped_ptr<CastRtpStream> stream2,
scoped_ptr<CastUdpTransport> udp_transport);
+ void CallStartCallback(int stream_id);
+ void CallStopCallback(int stream_id);
+ void CallErrorCallback(int stream_id, const std::string& message);
+
// Gets the RTP stream or UDP transport indexed by an ID.
// If not found, returns NULL and throws a V8 exception.
CastRtpStream* GetRtpStreamOrThrow(int stream_id) const;
diff --git a/chrome/renderer/extensions/chrome_v8_context.cc b/chrome/renderer/extensions/chrome_v8_context.cc
index 631a8b2..cb3b068 100644
--- a/chrome/renderer/extensions/chrome_v8_context.cc
+++ b/chrome/renderer/extensions/chrome_v8_context.cc
@@ -16,6 +16,7 @@
#include "content/public/renderer/v8_value_converter.h"
#include "extensions/common/extension.h"
#include "extensions/common/extension_api.h"
+#include "extensions/common/extension_urls.h"
#include "third_party/WebKit/public/web/WebFrame.h"
#include "third_party/WebKit/public/web/WebScopedMicrotaskSuppression.h"
#include "third_party/WebKit/public/web/WebView.h"
@@ -119,6 +120,17 @@ Feature::Availability ChromeV8Context::GetAvailability(
GetURL());
}
+void ChromeV8Context::DispatchEvent(const char* event_name,
+ v8::Handle<v8::Array> args) const {
+ v8::HandleScope handle_scope(isolate());
+ v8::Context::Scope context_scope(v8_context());
+
+ v8::Handle<v8::Value> argv[] = {
+ v8::String::NewFromUtf8(isolate(), event_name), args};
+ module_system_->CallModuleMethod(
+ kEventBindings, "dispatchEvent", arraysize(argv), argv);
+}
+
void ChromeV8Context::DispatchOnUnloadEvent() {
module_system_->CallModuleMethod("unload_event", "dispatch");
}
diff --git a/chrome/renderer/extensions/chrome_v8_context.h b/chrome/renderer/extensions/chrome_v8_context.h
index dcdcc901..ff44de8 100644
--- a/chrome/renderer/extensions/chrome_v8_context.h
+++ b/chrome/renderer/extensions/chrome_v8_context.h
@@ -99,6 +99,8 @@ class ChromeV8Context : public RequestSender::Source {
int argc,
v8::Handle<v8::Value> argv[]) const;
+ void DispatchEvent(const char* event_name, v8::Handle<v8::Array> args) const;
+
// Fires the onunload event on the unload_event module.
void DispatchOnUnloadEvent();
diff --git a/chrome/renderer/media/cast_rtp_stream.cc b/chrome/renderer/media/cast_rtp_stream.cc
index 06ffe7b..4824b5e 100644
--- a/chrome/renderer/media/cast_rtp_stream.cc
+++ b/chrome/renderer/media/cast_rtp_stream.cc
@@ -13,6 +13,7 @@
#include "content/public/renderer/media_stream_video_sink.h"
#include "content/public/renderer/render_thread.h"
#include "media/base/audio_bus.h"
+#include "media/base/bind_to_current_loop.h"
#include "media/cast/cast_config.h"
#include "media/cast/cast_defines.h"
#include "media/cast/cast_sender.h"
@@ -114,9 +115,13 @@ void DeleteAudioBus(scoped_ptr<media::AudioBus> audio_bus) {
class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>,
public content::MediaStreamVideoSink {
public:
- explicit CastVideoSink(const blink::WebMediaStreamTrack& track)
+ // |track| provides data for this sink.
+ // |error_callback| is called if video formats don't match.
+ CastVideoSink(const blink::WebMediaStreamTrack& track,
+ const CastRtpStream::ErrorCallback& error_callback)
: track_(track),
sink_added_(false),
+ error_callback_(error_callback),
render_thread_task_runner_(content::RenderThread::Get()
->GetMessageLoop()
->message_loop_proxy()) {}
@@ -153,6 +158,7 @@ class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>,
blink::WebMediaStreamTrack track_;
scoped_refptr<media::cast::FrameInput> frame_input_;
bool sink_added_;
+ CastRtpStream::ErrorCallback error_callback_;
scoped_refptr<base::SingleThreadTaskRunner> render_thread_task_runner_;
DISALLOW_COPY_AND_ASSIGN(CastVideoSink);
@@ -165,9 +171,13 @@ class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>,
class CastAudioSink : public base::SupportsWeakPtr<CastAudioSink>,
public content::MediaStreamAudioSink {
public:
- explicit CastAudioSink(const blink::WebMediaStreamTrack& track)
+ // |track| provides data for this sink.
+ // |error_callback| is called if audio formats don't match.
+ CastAudioSink(const blink::WebMediaStreamTrack& track,
+ const CastRtpStream::ErrorCallback& error_callback)
: track_(track),
sink_added_(false),
+ error_callback_(error_callback),
weak_factory_(this),
render_thread_task_runner_(content::RenderThread::Get()
->GetMessageLoop()
@@ -191,6 +201,7 @@ class CastAudioSink : public base::SupportsWeakPtr<CastAudioSink>,
// TODO(hclam): Pass in the accurate capture time to have good
// audio / video sync.
+ // TODO(hclam): We shouldn't hop through the render thread.
// Bounce the call from the real-time audio thread to the render thread.
// Needed since frame_input_ can be changed runtime by the render thread.
media::AudioBus* const audio_bus_ptr = audio_bus.get();
@@ -230,6 +241,7 @@ class CastAudioSink : public base::SupportsWeakPtr<CastAudioSink>,
blink::WebMediaStreamTrack track_;
scoped_refptr<media::cast::FrameInput> frame_input_;
bool sink_added_;
+ CastRtpStream::ErrorCallback error_callback_;
base::WeakPtrFactory<CastAudioSink> weak_factory_;
scoped_refptr<base::SingleThreadTaskRunner> render_thread_task_runner_;
@@ -264,12 +276,9 @@ CastRtpParams::CastRtpParams() {
CastRtpParams::~CastRtpParams() {
}
-CastRtpStream::CastRtpStream(
- const blink::WebMediaStreamTrack& track,
- const scoped_refptr<CastSession>& session)
- : track_(track),
- cast_session_(session) {
-}
+CastRtpStream::CastRtpStream(const blink::WebMediaStreamTrack& track,
+ const scoped_refptr<CastSession>& session)
+ : track_(track), cast_session_(session), weak_factory_(this) {}
CastRtpStream::~CastRtpStream() {
}
@@ -285,35 +294,60 @@ CastRtpParams CastRtpStream::GetParams() {
return params_;
}
-void CastRtpStream::Start(const CastRtpParams& params) {
+void CastRtpStream::Start(const CastRtpParams& params,
+ const base::Closure& start_callback,
+ const base::Closure& stop_callback,
+ const ErrorCallback& error_callback) {
+ stop_callback_ = stop_callback;
+ error_callback_ = error_callback;
+
if (IsAudio()) {
AudioSenderConfig config;
if (!ToAudioSenderConfig(params, &config)) {
- DVLOG(1) << "Invalid parameters for audio.";
+ DidEncounterError("Invalid parameters for audio.");
+ return;
}
- audio_sink_.reset(new CastAudioSink(track_));
+ // In case of error we have to go through DidEncounterError() to stop
+ // the streaming after reporting the error.
+ audio_sink_.reset(new CastAudioSink(
+ track_,
+ media::BindToCurrentLoop(base::Bind(&CastRtpStream::DidEncounterError,
+ weak_factory_.GetWeakPtr()))));
cast_session_->StartAudio(
config,
base::Bind(&CastAudioSink::AddToTrack,
audio_sink_->AsWeakPtr()));
+ start_callback.Run();
} else {
VideoSenderConfig config;
if (!ToVideoSenderConfig(params, &config)) {
- DVLOG(1) << "Invalid parameters for video.";
+ DidEncounterError("Invalid parameters for video.");
+ return;
}
- video_sink_.reset(new CastVideoSink(track_));
+ // See the code for audio above for explanation of callbacks.
+ video_sink_.reset(new CastVideoSink(
+ track_,
+ media::BindToCurrentLoop(base::Bind(&CastRtpStream::DidEncounterError,
+ weak_factory_.GetWeakPtr()))));
cast_session_->StartVideo(
config,
base::Bind(&CastVideoSink::AddToTrack,
video_sink_->AsWeakPtr()));
+ start_callback.Run();
}
}
void CastRtpStream::Stop() {
audio_sink_.reset();
video_sink_.reset();
+ stop_callback_.Run();
}
bool CastRtpStream::IsAudio() const {
return track_.source().type() == blink::WebMediaStreamSource::TypeAudio;
}
+
+void CastRtpStream::DidEncounterError(const std::string& message) {
+ error_callback_.Run(message);
+ Stop();
+}
diff --git a/chrome/renderer/media/cast_rtp_stream.h b/chrome/renderer/media/cast_rtp_stream.h
index a9fe5ed..f3a06c0 100644
--- a/chrome/renderer/media/cast_rtp_stream.h
+++ b/chrome/renderer/media/cast_rtp_stream.h
@@ -9,8 +9,10 @@
#include <vector>
#include "base/basictypes.h"
+#include "base/callback.h"
#include "base/memory/ref_counted.h"
#include "base/memory/scoped_ptr.h"
+#include "base/memory/weak_ptr.h"
#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
class CastAudioSink;
@@ -90,6 +92,8 @@ struct CastRtpParams {
// stream.
class CastRtpStream {
public:
+ typedef base::Callback<void(const std::string&)> ErrorCallback;
+
CastRtpStream(const blink::WebMediaStreamTrack& track,
const scoped_refptr<CastSession>& session);
~CastRtpStream();
@@ -102,7 +106,13 @@ class CastRtpStream {
// Begin encoding of media stream and then submit the encoded streams
// to underlying transport.
- void Start(const CastRtpParams& params);
+ // When the stream is started |start_callback| is called.
+ // When the stream is stopped |stop_callback| is called.
+ // When there is an error |error_callback| is called with a message.
+ void Start(const CastRtpParams& params,
+ const base::Closure& start_callback,
+ const base::Closure& stop_callback,
+ const ErrorCallback& error_callback);
// Stop encoding.
void Stop();
@@ -112,11 +122,16 @@ class CastRtpStream {
// track is a video track.
bool IsAudio() const;
+ void DidEncounterError(const std::string& message);
+
blink::WebMediaStreamTrack track_;
const scoped_refptr<CastSession> cast_session_;
scoped_ptr<CastAudioSink> audio_sink_;
scoped_ptr<CastVideoSink> video_sink_;
CastRtpParams params_;
+ base::WeakPtrFactory<CastRtpStream> weak_factory_;
+ base::Closure stop_callback_;
+ ErrorCallback error_callback_;
DISALLOW_COPY_AND_ASSIGN(CastRtpStream);
};
diff --git a/chrome/test/data/extensions/api_test/cast_streaming/basics.js b/chrome/test/data/extensions/api_test/cast_streaming/basics.js
index 3c8e410..d6d0bf4 100644
--- a/chrome/test/data/extensions/api_test/cast_streaming/basics.js
+++ b/chrome/test/data/extensions/api_test/cast_streaming/basics.js
@@ -8,6 +8,35 @@ var udpTransport = chrome.cast.streaming.udpTransport;
var createSession = chrome.cast.streaming.session.create;
var pass = chrome.test.callbackPass;
+function TestStateMachine(stream, audioId, videoId, udpId) {
+ this.stream = stream;
+ this.audioId = audioId;
+ this.videoId = videoId;
+ this.udpId = udpId;
+ this.audioStarted = false;
+ this.videoStarted = false;
+ this.audioStopped = false;
+ this.videoStopped = false;
+}
+
+TestStateMachine.prototype.onStarted = function(id) {
+ if (id == this.audioId)
+ this.audioStarted = true;
+ if (id == this.videoId)
+ this.videoStarted = true;
+ if (this.audioStarted && this.videoStarted)
+ this.onAllStarted();
+}
+
+TestStateMachine.prototype.onStopped = function(id) {
+ if (id == this.audioId)
+ this.audioStopped = true;
+ if (id == this.videoId)
+ this.videoStopped = true;
+ if (this.audioStopped && this.videoStopped)
+ this.onAllStopped();
+}
+
chrome.test.runTests([
function rtpStreamStart() {
console.log("[TEST] rtpStreamStart");
@@ -19,23 +48,38 @@ chrome.test.runTests([
stream.getVideoTracks()[0],
pass(function(stream, audioId, videoId, udpId) {
console.log("Starting.");
+ var stateMachine = new TestStateMachine(stream,
+ audioId,
+ videoId,
+ udpId);
var audioParams = rtpStream.getSupportedParams(audioId)[0];
var videoParams = rtpStream.getSupportedParams(videoId)[0];
+ chrome.test.assertEq(audioParams.payload.codecName, "OPUS");
+ chrome.test.assertEq(videoParams.payload.codecName, "VP8");
udpTransport.setDestination(udpId,
{address: "127.0.0.1", port: 2344});
- rtpStream.start(audioId, audioParams);
- rtpStream.start(videoId, videoParams);
- window.setTimeout(pass(function() {
+ rtpStream.onStarted.addListener(
+ stateMachine.onStarted.bind(stateMachine));
+ stateMachine.onAllStarted =
+ pass(function(audioId, videoId) {
console.log("Stopping.");
rtpStream.stop(audioId);
rtpStream.stop(videoId);
+ }.bind(null, audioId, videoId));
+ rtpStream.onStopped.addListener(
+ stateMachine.onStopped.bind(stateMachine));
+ stateMachine.onAllStopped =
+ pass(function(stream, audioId, videoId, udpId) {
+ console.log("Destroying.");
rtpStream.destroy(audioId);
rtpStream.destroy(videoId);
udpTransport.destroy(udpId);
chrome.test.assertEq(audioParams.payload.codecName, "OPUS");
chrome.test.assertEq(videoParams.payload.codecName, "VP8");
chrome.test.succeed();
- }), 0);
+ }.bind(null, stream, audioId, videoId, udpId));
+ rtpStream.start(audioId, audioParams);
+ rtpStream.start(videoId, videoParams);
}.bind(null, stream)));
}));
},
generated by cgit v1.1 at 2025-02-04 03:04:21 +0000