| 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;
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| - std::string value;
|
| -
|
| - CastCodecSpecificParams();
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| - ~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.
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| - int min_bitrate;
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| -
|
| - // 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();
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| - ~CastRtpPayloadParams();
|
| -};
|
| -
|
| -// Defines the capabilities of the transport.
|
| -struct CastRtpCaps {
|
| - // Defines a list of supported payloads.
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| - std::vector<CastRtpPayloadParams> payloads;
|
| -
|
| - // Names of supported RTCP features.
|
| - std::vector<std::string> rtcp_features;
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| -
|
| - // 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);
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| - ~CastSendTransport();
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| -
|
| - // Return capabilities currently supported by this transport.
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| - CastRtpCaps GetCaps();
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| -
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| - // Return parameters set to this transport.
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| - CastRtpParams GetParams();
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| -
|
| - // Begin encoding of media stream and then submit the encoded streams
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| - // to underlying transport.
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| - void Start(const CastRtpParams& params);
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| -
|
| - // Stop encoding.
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| - void Stop();
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| -
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| - private:
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| - // Return true if this track is an audio track. Return false if this
|
| - // track is a video track.
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| - bool IsAudio() const;
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| -
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| - blink::WebMediaStreamTrack track_;
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| - const scoped_refptr<CastSession> cast_session_;
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| - CastRtpParams params_;
|
| -
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| - DISALLOW_COPY_AND_ASSIGN(CastSendTransport);
|
| -};
|
| -
|
| -#endif // CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_
|
|
|