Define Cast Content API

content/public/renderer/webrtc_cast_send_transport.h

+// 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 CONTENT_PUBLIC_RENDERER_MEDIA_WEBRTC_CAST_SEND_TRANSPORT_H_

[jam, 2013/10/11 16:47:14]

nit: get rid of _MEDIA_

[Alpha Left Google, 2013/10/11 20:24:21]

Done.

+#define CONTENT_PUBLIC_RENDERER_MEDIA_WEBRTC_CAST_SEND_TRANSPORT_H_
+
+#include <string>
+#include <vector>
+
+#include "base/callback.h"

[jam, 2013/10/11 16:47:14]

nit: not needed

[Alpha Left Google, 2013/10/11 20:24:21]

Done.

+#include "content/common/content_export.h"
+
+namespace WebKit {
+class WebMediaStreamTrack;
+}  // namespace WebKit
+
+namespace content {
+
+class WebRtcUdpTransport;
+
+// A key value pair structure for codec specific parameters.
+struct CONTENT_EXPORT WebRtcCodecSpecificParam {

[jam, 2013/10/11 16:47:14]

this isn't used?

[Alpha Left Google, 2013/10/11 20:24:21]

Thanks. I forgot to use this WebRtcRtpPayloadParam. There should be a list
there. Added in the latest patch.

+  std::string key;
+  std::string value;
+
+  WebRtcCodecSpecificParam();
+  ~WebRtcCodecSpecificParam();
+};
+
+// Defines the basic properties of a payload supported by cast transport.
+struct CONTENT_EXPORT WebRtcRtpPayloadParam {
+  // 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;
+
+  WebRtcRtpPayloadParam();

[jam, 2013/10/11 16:47:14]

can you inline this without getting a clang error?

[Alpha Left Google, 2013/10/11 20:24:21]

I tried but clang complained that I should put the ctors and dtors in a separate
.cc file.

For example:
https://code.google.com/p/chromium/codesearch#chromium/src/content/public/ren...

+  ~WebRtcRtpPayloadParam();
+};
+
+// Defines the capabilities of the transport.
+struct CONTENT_EXPORT WebRtcRtpCaps {
+  // Defines a list of supported payloads.
+  std::vector<WebRtcRtpPayloadParam> 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;
+
+  WebRtcRtpCaps();
+  ~WebRtcRtpCaps();
+};
+
+typedef WebRtcRtpCaps WebRtcRtpParams;
+
+// This interface defines operations needed by an application to use the
+// Cast transport protocol and interact with WebRTC objects.
+// This interface takes input from a WebMediaStreamTrack which contains
+// audio or video stream and then send the encoded stream to the
+// underlying transport, e.g. a UDP transport.
+class CONTENT_EXPORT WebRtcCastSendTransport {

[jam, 2013/10/11 16:47:14]

nit: export not needed

[Alpha Left Google, 2013/10/11 20:24:21]

Done.

+ public:
+  // Return capabilities currently spported by this transport.
+  virtual WebRtcRtpCaps getCaps() = 0;

[jam, 2013/10/11 16:47:14]

nit: please check the google style guide

[Alpha Left Google, 2013/10/11 20:24:21]

Oooops sorry.. Been doing too much blink lately.

+
+  // Return parameters set to this transport.
+  virtual WebRtcRtpParams getParams() = 0;
+
+  // Return the best parameters given the capabilities of remote peer.
+  virtual WebRtcRtpParams createParams(WebRtcRtpCaps remote_caps) = 0;
+
+  // Begin encoding of media stream from |track| and submit the encoded
+  // stream to underlying transport.
+  virtual void start(
+      WebKit::WebMediaStreamTrack* track,
+      WebRtcRtpParams params) = 0;
+
+  // Stop encoding.
+  virtual void stop() = 0;
+
+ protected:
+  virtual ~WebRtcCastSendTransport() {}
+};
+
+// Create a Cast send transport providing the underlying UDP transport.
+CONTENT_EXPORT WebRtcCastSendTransport* CreateWebRtcCastSendTransport(

[jam, 2013/10/11 16:47:14]

nit: put this as a static method in WebRtcCastSendTransport called Create (by
convention)

[Alpha Left Google, 2013/10/11 20:24:21]

Done.

+    WebRtcUdpTransport* udp_transport);
+
+}  // namespace content
+
+#endif  // CONTENT_PUBLIC_RENDERER_MEDIA_WEBRTC_CAST_SEND_TRANSPORT_H_