Refactoring: Merge VideoSenderConfig and AudioSenderConfig.

chrome/renderer/media/cast_rtp_stream.h

Index: chrome/renderer/media/cast_rtp_stream.h
diff --git a/chrome/renderer/media/cast_rtp_stream.h b/chrome/renderer/media/cast_rtp_stream.h
index e0ecdf735173a08fc75ac3df17847591d7bd523f..01c3da8e0e815f2c40ccda6885ef18ca236e96bd 100644
--- a/chrome/renderer/media/cast_rtp_stream.h
+++ b/chrome/renderer/media/cast_rtp_stream.h
@@ -26,85 +26,6 @@ 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 {

[miu, 2016/06/30 21:59:43]

Nice! :)

[xjz, 2016/07/01 23:52:09]

:)

-  // RTP specific field that identifies the content type.
-  int payload_type = media::cast::kDefaultRtpVideoPayloadType;
-
-  // 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
@@ -114,22 +35,23 @@ 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);
@@ -162,7 +84,6 @@ class CastRtpStream {
   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_;