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      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   // Width and height of the video content.
     71   int width;
     72   int height;
     73 
     74   // Name of the codec used.
     75   std::string codec_name;
     76 
     77   // AES encryption key.
     78   std::string aes_key;
     79 
     80   // AES encryption IV mask.
     81   std::string aes_iv_mask;
     82 
     83   // List of codec specific parameters.
     84   std::vector<CastCodecSpecificParams> codec_specific_params;
     85 
     86   CastRtpPayloadParams();
     87   ~CastRtpPayloadParams();
     88 };
     89 
     90 // Defines the parameters of a RTP stream.
     91 struct CastRtpParams {
     92   explicit CastRtpParams(const CastRtpPayloadParams& payload_params);
     93 
     94   // Payload parameters.
     95   CastRtpPayloadParams payload;
     96 
     97   // Names of supported RTCP features.
     98   std::vector<std::string> rtcp_features;
     99 
    100   CastRtpParams();
    101   ~CastRtpParams();
    102 };
    103 
    104 // This object represents a RTP stream that encodes and optionally
    105 // encrypt audio or video data from a WebMediaStreamTrack.
    106 // Note that this object does not actually output packets. It allows
    107 // configuration of encoding and RTP parameters and control such a logical
    108 // stream.
    109 class CastRtpStream {
    110  public:
    111   typedef base::Callback<void(const std::string&)> ErrorCallback;
    112 
    113   CastRtpStream(const blink::WebMediaStreamTrack& track,
    114                 const scoped_refptr<CastSession>& session);
    115   ~CastRtpStream();
    116 
    117   // Return parameters currently supported by this stream.
    118   std::vector<CastRtpParams> GetSupportedParams();
    119 
    120   // Return parameters set to this stream.
    121   CastRtpParams GetParams();
    122 
    123   // Begin encoding of media stream and then submit the encoded streams
    124   // to underlying transport.
    125   // When the stream is started |start_callback| is called.
    126   // When the stream is stopped |stop_callback| is called.
    127   // When there is an error |error_callback| is called with a message.
    128   void Start(const CastRtpParams& params,
    129              const base::Closure& start_callback,
    130              const base::Closure& stop_callback,
    131              const ErrorCallback& error_callback);
    132 
    133   // Stop encoding.
    134   void Stop();
    135 
    136   // Enables or disables logging for this stream.
    137   void ToggleLogging(bool enable);
    138 
    139   // Get serialized raw events for this stream with |extra_data| attached,
    140   // and invokes |callback| with the result.
    141   void GetRawEvents(
    142       const base::Callback<void(scoped_ptr<base::BinaryValue>)>& callback,
    143       const std::string& extra_data);
    144 
    145   // Get stats in DictionaryValue format and invokves |callback| with
    146   // the result.
    147   void GetStats(const base::Callback<void(
    148       scoped_ptr<base::DictionaryValue>)>& callback);
    149 
    150  private:
    151   // Return true if this track is an audio track. Return false if this
    152   // track is a video track.
    153   bool IsAudio() const;
    154 
    155   void DidEncounterError(const std::string& message);
    156 
    157   blink::WebMediaStreamTrack track_;
    158   const scoped_refptr<CastSession> cast_session_;
    159   scoped_ptr<CastAudioSink> audio_sink_;
    160   scoped_ptr<CastVideoSink> video_sink_;
    161   CastRtpParams params_;
    162   base::Closure stop_callback_;
    163   ErrorCallback error_callback_;
    164 
    165   base::WeakPtrFactory<CastRtpStream> weak_factory_;
    166 
    167   DISALLOW_COPY_AND_ASSIGN(CastRtpStream);
    168 };
    169 
    170 #endif  // CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
    171