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Unified Diff: chrome/renderer/media/cast_send_transport.h

Issue 26931002: Define Cast Content API (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Created 7 years, 2 months ago
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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
new file mode 100644
index 0000000000000000000000000000000000000000..c8e4bcf8fe1c04099d608c99c9045a9f2eb63bd8
--- /dev/null
+++ b/chrome/renderer/media/cast_send_transport.h
@@ -0,0 +1,109 @@
+// 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"
+
+namespace WebKit {
+class WebMediaStreamTrack;
+} // namespace WebKit
+
+class CastUdpTransport;
+
+// A key value pair structure for codec specific parameters.
+struct CastCodecSpecificParam {
+ std::string key;
+ std::string value;
+
+ CastCodecSpecificParam();
+ ~CastCodecSpecificParam();
+};
+
+// Defines the basic properties of a payload supported by cast transport.
+struct CastRtpPayloadParam {
+ // 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;
+
+ // Uncompressed bitrate.
+ int 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<CastCodecSpecificParam> codec_specific_params;
+
+ CastRtpPayloadParam();
+ ~CastRtpPayloadParam();
+};
+
+// Defines the capabilities of the transport.
+struct CastRtpCaps {
+ // Defines a list of supported payloads.
+ std::vector<CastRtpPayloadParam> 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_mechanism;
+
+ CastRtpCaps();
+ ~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
+// stream.
+class CastSendTransport {
+ public:
+ CastSendTransport(CastUdpTransport* udp_transport);
+ virtual ~CastSendTransport();
+
+ // Return capabilities currently spported by this transport.
+ CastRtpCaps GetCaps();
+
+ // Return parameters set to this transport.
+ CastRtpParams GetParams();
+
+ // Return the best parameters given the capabilities of remote peer.
+ CastRtpParams CreateParams(CastRtpCaps remote_caps);
+
+ // Begin encoding of media stream from |audio_track| and |video_track|
+ // and then submit the encoded streams to underlying transport.
+ // Either stream can be NULL but it is invalid for both streams to be
+ // NULL.
+ void Start(WebKit::WebMediaStreamTrack* audio_track,
+ WebKit::WebMediaStreamTrack* video_track,
+ CastRtpParams params);
+
+ // Stop encoding.
+ void Stop();
+
+ private:
+ DISALLOW_COPY_AND_ASSIGN(CastSendTransport);
+};
+
+#endif // CHROME_RENDERER_MEDIA_CAST_SEND_TRANSPORT_H_
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