Refactoring: Merge VideoSenderConfig and AudioSenderConfig.

chrome/renderer/media/cast_rtp_stream.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 CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
 #define CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "base/callback.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/weak_ptr.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/constants.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 
 namespace base {
 class BinaryValue;
 class DictionaryValue;
 }
 
 class CastAudioSink;
 class CastSession;
 class CastVideoSink;
 
-// A key value pair structure for codec specific parameters.
-struct CastCodecSpecificParams {
-  std::string key;
-  std::string value;
-
-  CastCodecSpecificParams();
-  ~CastCodecSpecificParams();
-};
-
-// Defines the basic properties of a payload supported by cast transport.
-struct CastRtpPayloadParams {
-  // RTP specific field that identifies the content type.
-  media::cast::RtpPayloadType payload_type =
-      media::cast::RtpPayloadType::VIDEO_VP8;
-
-  // Maximum latency in milliseconds. Implemetation tries to keep latency
-  // under this threshold.
-  int max_latency_ms = media::cast::kDefaultRtpMaxDelayMs;
-
-  // Minimum latency.
-  // Default value (0) means use max_latency_ms.
-  int min_latency_ms = 0;
-
-  // Starting latency on animated content.
-  // Default value (0) means use max_latency_ms.
-  int animated_latency_ms = 0;
-
-  // RTP specific field to identify a stream.
-  int ssrc = 1;
-
-  // RTP specific field to idenfity the feedback stream.
-  int feedback_ssrc = 2;
-
-  // Update frequency of payload sample.
-  int clock_rate = media::cast::kVideoFrequency;
-
-  // Maximum bitrate in kilobits per second.
-  int max_bitrate = media::cast::kDefaultMaxVideoKbps;
-
-  // Minimum bitrate in kilobits per second.
-  int min_bitrate = media::cast::kDefaultMinVideoKbps;
-
-  // Number of audio channels.
-  int channels = 1;
-
-  // The maximum frame rate.
-  double max_frame_rate = media::cast::kDefaultMaxFrameRate;
-
-  // Name of the codec used.
-  std::string codec_name;
-
-  // AES encryption key.
-  std::string aes_key;
-
-  // AES encryption IV mask.
-  std::string aes_iv_mask;
-
-  // List of codec specific parameters.
-  std::vector<CastCodecSpecificParams> codec_specific_params;
-
-  CastRtpPayloadParams();
-  CastRtpPayloadParams(const CastRtpPayloadParams& other);
-  ~CastRtpPayloadParams();
-};
-
-// Defines the parameters of a RTP stream.
-struct CastRtpParams {
-  explicit CastRtpParams(const CastRtpPayloadParams& payload_params);
-
-  // Payload parameters.
-  CastRtpPayloadParams payload;
-
-  // Names of supported RTCP features.
-  std::vector<std::string> rtcp_features;
-
-  CastRtpParams();
-  CastRtpParams(const CastRtpParams& other);
-  ~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 CastRtpStream {
  public:
   typedef base::Callback<void(const std::string&)> ErrorCallback;
 
+  static bool IsHardwareVP8EncodingSupported();
+
+  static bool IsHardwareH264EncodingSupported();
+
   CastRtpStream(const blink::WebMediaStreamTrack& track,
                 const scoped_refptr<CastSession>& session);
   ~CastRtpStream();
 
   // Return parameters currently supported by this stream.
-  std::vector<CastRtpParams> GetSupportedParams();
-
-  // Return parameters set to this stream.
-  CastRtpParams GetParams();
+  std::vector<media::cast::FrameSenderConfig> GetSupportedConfigs();
 
   // Begin encoding of media stream and then submit the encoded streams
   // to underlying transport.
   // When the stream is started |start_callback| is called.
   // When the stream is stopped |stop_callback| is called.
   // When there is an error |error_callback| is called with a message.
-  void Start(const CastRtpParams& params,
+  void Start(const media::cast::FrameSenderConfig& config,
              const base::Closure& start_callback,
              const base::Closure& stop_callback,
              const ErrorCallback& error_callback);
 
   // Stop encoding.
   void Stop();
 
   // Enables or disables logging for this stream.
   void ToggleLogging(bool enable);
 
@@ skipping 12 matching lines @@
   // Return true if this track is an audio track. Return false if this
   // track is a video track.
   bool IsAudio() const;
 
   void DidEncounterError(const std::string& message);
 
   blink::WebMediaStreamTrack track_;
   const scoped_refptr<CastSession> cast_session_;
   std::unique_ptr<CastAudioSink> audio_sink_;
   std::unique_ptr<CastVideoSink> video_sink_;
-  CastRtpParams params_;
   base::Closure stop_callback_;
   ErrorCallback error_callback_;
 
   base::WeakPtrFactory<CastRtpStream> weak_factory_;
 
   DISALLOW_COPY_AND_ASSIGN(CastRtpStream);
 };
 
 #endif  // CHROME_RENDERER_MEDIA_CAST_RTP_STREAM_H_