| Index: chrome/renderer/media/cast_send_transport.h
|
| diff --git a/chrome/renderer/media/cast_send_transport.h b/chrome/renderer/media/cast_send_transport.h
|
| index 1b3c64ab5254bf921f70ab398a70bf4e6113fd65..b30f564f216ac9f7c1629f4bcc2294ef6ff48b30 100644
|
| --- a/chrome/renderer/media/cast_send_transport.h
|
| +++ b/chrome/renderer/media/cast_send_transport.h
|
| @@ -13,7 +13,6 @@
|
| #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
|
|
|
| class CastSession;
|
| -class CastUdpTransport;
|
|
|
| // A key value pair structure for codec specific parameters.
|
| struct CastCodecSpecificParams {
|
| @@ -75,14 +74,15 @@ struct CastRtpCaps {
|
|
|
| typedef CastRtpCaps CastRtpParams;
|
|
|
| -// This class takes input from audio and/or video WebMediaStreamTracks
|
| -// and then send the encoded streams to the underlying transport,
|
| -// e.g. a UDP transport. It also allows configuration of the encoded
|
| +// This object represents a RTP stream that encodes and optionally
|
| +// encrypt audio or video data from a WebMediaStreamTrack.
|
| +// Note that this object does not actually output packets. It allows
|
| +// configuration of encoding and RTP parameters and control such a logical
|
| // stream.
|
| class CastSendTransport {
|
| public:
|
| - CastSendTransport(CastUdpTransport* udp_transport,
|
| - const blink::WebMediaStreamTrack& track);
|
| + CastSendTransport(const blink::WebMediaStreamTrack& track,
|
| + const scoped_refptr<CastSession>& session);
|
| ~CastSendTransport();
|
|
|
| // Return capabilities currently supported by this transport.
|
| @@ -103,8 +103,8 @@ class CastSendTransport {
|
| // track is a video track.
|
| bool IsAudio() const;
|
|
|
| - const scoped_refptr<CastSession> cast_session_;
|
| blink::WebMediaStreamTrack track_;
|
| + const scoped_refptr<CastSession> cast_session_;
|
| CastRtpParams params_;
|
|
|
| DISALLOW_COPY_AND_ASSIGN(CastSendTransport);
|
|
|