chrome/renderer/media/cast_rtp_stream.h

   1 // Copyright 2013 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #ifndef CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
   6 #define CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
   7
   8 #include <string>
   9 #include <vector>
  10
  11 #include "base/basictypes.h"
  12 #include "base/callback.h"
  13 #include "base/memory/ref_counted.h"
  14 #include "base/memory/scoped_ptr.h"
  15 #include "base/memory/weak_ptr.h"
  16 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  17
  18 namespace base {
  19 class BinaryValue;
  20 class DictionaryValue;
  21 }
  22
  23 class CastAudioSink;
  24 class CastSession;
  25 class CastVideoSink;
  26
  27 // A key value pair structure for codec specific parameters.
  28 struct CastCodecSpecificParams {
  29   std::string key;
  30   std::string value;
  31
  32   CastCodecSpecificParams();
  33   ~CastCodecSpecificParams();
  34 };
  35
  36 // Defines the basic properties of a payload supported by cast transport.
  37 struct CastRtpPayloadParams {
  38   // RTP specific field that identifies the content type.
  39   int payload_type;
  40
  41   // Maximum latency in milliseconds. Implemetation tries to keep latency
  42   // under this threshold.
  43   int max_latency_ms;
  44
  45   // Minimum latency.
  46   // Default value (0) means use max_latency_ms.
  47   int min_latency_ms;
  48
  49   // RTP specific field to identify a stream.
  50   int ssrc;
  51
  52   // RTP specific field to idenfity the feedback stream.
  53   int feedback_ssrc;
  54
  55   // Update frequency of payload sample.
  56   int clock_rate;
  57
  58   // Maximum bitrate in kilobits per second.
  59   int max_bitrate;
  60
  61   // Minimum bitrate in kilobits per second.
  62   int min_bitrate;
  63
  64   // Number of audio channels.
  65   int channels;
  66
  67   // The maximum frame rate.
  68   double max_frame_rate;
  69
  70   // Name of the codec used.
  71   std::string codec_name;
  72
  73   // AES encryption key.
  74   std::string aes_key;
  75
  76   // AES encryption IV mask.
  77   std::string aes_iv_mask;
  78
  79   // List of codec specific parameters.
  80   std::vector<CastCodecSpecificParams> codec_specific_params;
  81
  82   CastRtpPayloadParams();
  83   ~CastRtpPayloadParams();
  84 };
  85
  86 // Defines the parameters of a RTP stream.
  87 struct CastRtpParams {
  88   explicit CastRtpParams(const CastRtpPayloadParams& payload_params);
  89
  90   // Payload parameters.
  91   CastRtpPayloadParams payload;
  92
  93   // Names of supported RTCP features.
  94   std::vector<std::string> rtcp_features;
  95
  96   CastRtpParams();
  97   ~CastRtpParams();
  98 };
  99
 100 // This object represents a RTP stream that encodes and optionally
 101 // encrypt audio or video data from a WebMediaStreamTrack.
 102 // Note that this object does not actually output packets. It allows
 103 // configuration of encoding and RTP parameters and control such a logical
 104 // stream.
 105 class CastRtpStream {
 106  public:
 107   typedef base::Callback<void(const std::string&)> ErrorCallback;
 108
 109   CastRtpStream(const blink::WebMediaStreamTrack& track,
 110                 const scoped_refptr<CastSession>& session);
 111   ~CastRtpStream();
 112
 113   // Return parameters currently supported by this stream.
 114   std::vector<CastRtpParams> GetSupportedParams();
 115
 116   // Return parameters set to this stream.
 117   CastRtpParams GetParams();
 118
 119   // Begin encoding of media stream and then submit the encoded streams
 120   // to underlying transport.
 121   // When the stream is started |start_callback| is called.
 122   // When the stream is stopped |stop_callback| is called.
 123   // When there is an error |error_callback| is called with a message.
 124   void Start(const CastRtpParams& params,
 125              const base::Closure& start_callback,
 126              const base::Closure& stop_callback,
 127              const ErrorCallback& error_callback);
 128
 129   // Stop encoding.
 130   void Stop();
 131
 132   // Enables or disables logging for this stream.
 133   void ToggleLogging(bool enable);
 134
 135   // Get serialized raw events for this stream with |extra_data| attached,
 136   // and invokes |callback| with the result.
 137   void GetRawEvents(
 138       const base::Callback<void(scoped_ptr<base::BinaryValue>)>& callback,
 139       const std::string& extra_data);
 140
 141   // Get stats in DictionaryValue format and invokves |callback| with
 142   // the result.
 143   void GetStats(const base::Callback<void(
 144       scoped_ptr<base::DictionaryValue>)>& callback);
 145
 146  private:
 147   // Return true if this track is an audio track. Return false if this
 148   // track is a video track.
 149   bool IsAudio() const;
 150
 151   void DidEncounterError(const std::string& message);
 152
 153   blink::WebMediaStreamTrack track_;
 154   const scoped_refptr<CastSession> cast_session_;
 155   scoped_ptr<CastAudioSink> audio_sink_;
 156   scoped_ptr<CastVideoSink> video_sink_;
 157   CastRtpParams params_;
 158   base::Closure stop_callback_;
 159   ErrorCallback error_callback_;
 160
 161   base::WeakPtrFactory<CastRtpStream> weak_factory_;
 162
 163   DISALLOW_COPY_AND_ASSIGN(CastRtpStream);
 164 };
 165
 166 #endif  // CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_