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 |
deleted file mode 100644 |
index b30f564f216ac9f7c1629f4bcc2294ef6ff48b30..0000000000000000000000000000000000000000 |
--- a/chrome/renderer/media/cast_send_transport.h |
+++ /dev/null |
@@ -1,113 +0,0 @@ |
-// Copyright 2013 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. |
- |
-#ifndef CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_ |
-#define CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_ |
- |
-#include <string> |
-#include <vector> |
- |
-#include "base/basictypes.h" |
-#include "base/memory/ref_counted.h" |
-#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
- |
-class CastSession; |
- |
-// A key value pair structure for codec specific parameters. |
-struct CastCodecSpecificParams { |
- std::string key; |
- std::string value; |
- |
- CastCodecSpecificParams(); |
- ~CastCodecSpecificParams(); |
-}; |
- |
-// Defines the basic properties of a payload supported by cast transport. |
-struct CastRtpPayloadParams { |
- // RTP specific field that identifies the content type. |
- int payload_type; |
- |
- // RTP specific field to identify a stream. |
- int ssrc; |
- |
- // Update frequency of payload sample. |
- int clock_rate; |
- |
- // Maximum bitrate. |
- int max_bitrate; |
- |
- // Minimum bitrate. |
- int min_bitrate; |
- |
- // Number of audio channels. |
- int channels; |
- |
- // Width and height of the video content. |
- int width; |
- int height; |
- |
- // Name of the codec used. |
- std::string codec_name; |
- |
- // List of codec specific parameters. |
- std::vector<CastCodecSpecificParams> codec_specific_params; |
- |
- CastRtpPayloadParams(); |
- ~CastRtpPayloadParams(); |
-}; |
- |
-// Defines the capabilities of the transport. |
-struct CastRtpCaps { |
- // Defines a list of supported payloads. |
- std::vector<CastRtpPayloadParams> payloads; |
- |
- // Names of supported RTCP features. |
- std::vector<std::string> rtcp_features; |
- |
- // Names of supported FEC (Forward Error Correction) mechanisms. |
- std::vector<std::string> fec_mechanisms; |
- |
- CastRtpCaps(); |
- ~CastRtpCaps(); |
-}; |
- |
-typedef CastRtpCaps CastRtpParams; |
- |
-// 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(const blink::WebMediaStreamTrack& track, |
- const scoped_refptr<CastSession>& session); |
- ~CastSendTransport(); |
- |
- // Return capabilities currently supported by this transport. |
- CastRtpCaps GetCaps(); |
- |
- // Return parameters set to this transport. |
- CastRtpParams GetParams(); |
- |
- // Begin encoding of media stream and then submit the encoded streams |
- // to underlying transport. |
- void Start(const CastRtpParams& params); |
- |
- // Stop encoding. |
- void Stop(); |
- |
- private: |
- // Return true if this track is an audio track. Return false if this |
- // track is a video track. |
- bool IsAudio() const; |
- |
- blink::WebMediaStreamTrack track_; |
- const scoped_refptr<CastSession> cast_session_; |
- CastRtpParams params_; |
- |
- DISALLOW_COPY_AND_ASSIGN(CastSendTransport); |
-}; |
- |
-#endif // CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_ |