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-rw-r--r--chrome/renderer/media/cast_rtp_stream.cc22
-rw-r--r--chrome/renderer/media/cast_rtp_stream.h1
-rw-r--r--chrome/test/data/extensions/api_test/cast_streaming/rtp_stream_error.js5
3 files changed, 3 insertions, 25 deletions
diff --git a/chrome/renderer/media/cast_rtp_stream.cc b/chrome/renderer/media/cast_rtp_stream.cc
index 4d51889..a3f970d 100644
--- a/chrome/renderer/media/cast_rtp_stream.cc
+++ b/chrome/renderer/media/cast_rtp_stream.cc
@@ -214,10 +214,6 @@ bool ToVideoSenderConfig(const CastRtpParams& params,
if (config->min_playout_delay > config->max_playout_delay)
return false;
config->rtp_payload_type = params.payload.payload_type;
- config->width = params.payload.width;
- config->height = params.payload.height;
- if (config->width < 2 || config->height < 2)
- return false;
config->min_bitrate = config->start_bitrate =
params.payload.min_bitrate * kBitrateMultiplier;
config->max_bitrate = params.payload.max_bitrate * kBitrateMultiplier;
@@ -257,14 +253,11 @@ class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>,
public content::MediaStreamVideoSink {
public:
// |track| provides data for this sink.
- // |expected_natural_size| is the expected dimension of the video frame.
// |error_callback| is called if video formats don't match.
CastVideoSink(const blink::WebMediaStreamTrack& track,
- const gfx::Size& expected_natural_size,
const CastRtpStream::ErrorCallback& error_callback)
: track_(track),
sink_added_(false),
- expected_natural_size_(expected_natural_size),
error_callback_(error_callback) {}
~CastVideoSink() override {
@@ -275,24 +268,12 @@ class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>,
// This static method is used to forward video frames to |frame_input|.
static void OnVideoFrame(
// These parameters are already bound when callback is created.
- const gfx::Size& expected_natural_size,
const CastRtpStream::ErrorCallback& error_callback,
const scoped_refptr<media::cast::VideoFrameInput> frame_input,
// These parameters are passed for each frame.
const scoped_refptr<media::VideoFrame>& frame,
const media::VideoCaptureFormat& format,
const base::TimeTicks& estimated_capture_time) {
- if (frame->natural_size() != expected_natural_size) {
- error_callback.Run(
- base::StringPrintf("Video frame resolution does not match config."
- " Expected %dx%d. Got %dx%d.",
- expected_natural_size.width(),
- expected_natural_size.height(),
- frame->natural_size().width(),
- frame->natural_size().height()));
- return;
- }
-
base::TimeTicks timestamp;
if (estimated_capture_time.is_null())
timestamp = base::TimeTicks::Now();
@@ -318,7 +299,6 @@ class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>,
this,
base::Bind(
&CastVideoSink::OnVideoFrame,
- expected_natural_size_,
error_callback_,
frame_input),
track_);
@@ -327,7 +307,6 @@ class CastVideoSink : public base::SupportsWeakPtr<CastVideoSink>,
private:
blink::WebMediaStreamTrack track_;
bool sink_added_;
- gfx::Size expected_natural_size_;
CastRtpStream::ErrorCallback error_callback_;
DISALLOW_COPY_AND_ASSIGN(CastVideoSink);
@@ -543,7 +522,6 @@ void CastRtpStream::Start(const CastRtpParams& params,
// See the code for audio above for explanation of callbacks.
video_sink_.reset(new CastVideoSink(
track_,
- gfx::Size(config.width, config.height),
media::BindToCurrentLoop(base::Bind(&CastRtpStream::DidEncounterError,
weak_factory_.GetWeakPtr()))));
cast_session_->StartVideo(
diff --git a/chrome/renderer/media/cast_rtp_stream.h b/chrome/renderer/media/cast_rtp_stream.h
index 24bf885..61a6bf4 100644
--- a/chrome/renderer/media/cast_rtp_stream.h
+++ b/chrome/renderer/media/cast_rtp_stream.h
@@ -68,6 +68,7 @@ struct CastRtpPayloadParams {
double max_frame_rate;
// Width and height of the video content.
+ // TODO(miu): DEPRECATED. Remove these, as they are ignored.
int width;
int height;
diff --git a/chrome/test/data/extensions/api_test/cast_streaming/rtp_stream_error.js b/chrome/test/data/extensions/api_test/cast_streaming/rtp_stream_error.js
index 5cdba8a..9db1949 100644
--- a/chrome/test/data/extensions/api_test/cast_streaming/rtp_stream_error.js
+++ b/chrome/test/data/extensions/api_test/cast_streaming/rtp_stream_error.js
@@ -39,9 +39,8 @@ chrome.test.runTests([
udpTransport.destroy(udpId);
console.log(msg);
}.bind(null, audioId, videoId)));
- // Invalid width and height.
- videoParams.payload.width = 100;
- videoParams.payload.height = 100;
+ // Specify invalid value to trigger error.
+ videoParams.payload.codecName = "Animated WebP";
udpTransport.setDestination(udpId,
{address: "127.0.0.1", port: 2344});
rtpStream.start(videoId, videoParams);