X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=src%2Fmedia%2Fcast%2Ftransport%2Fcast_transport_config.h;h=d8474f40d66656bf84fc5d2211d4625de3d96f74;hb=ff3e2503a20db9193d323c1d19c38c68004dec4a;hp=15015cdba06f94da5aa4b8a9763647785e1d9dbf;hpb=7338fba38ba696536d1cc9d389afd716a6ab2fe6;p=platform%2Fframework%2Fweb%2Fcrosswalk.git

diff --git a/src/media/cast/transport/cast_transport_config.h b/src/media/cast/transport/cast_transport_config.h
index 15015cd..d8474f4 100644
--- a/src/media/cast/transport/cast_transport_config.h
+++ b/src/media/cast/transport/cast_transport_config.h
@@ -12,25 +12,26 @@
 #include "base/callback.h"
 #include "base/memory/ref_counted.h"
 #include "media/cast/transport/cast_transport_defines.h"
+#include "net/base/ip_endpoint.h"
 
 namespace media {
 namespace cast {
 namespace transport {
 
 enum RtcpMode {
-  kRtcpCompound,  // Compound RTCP mode is described by RFC 4585.
+  kRtcpCompound,     // Compound RTCP mode is described by RFC 4585.
   kRtcpReducedSize,  // Reduced-size RTCP mode is described by RFC 5506.
 };
 
-enum VideoCodec {
-  kVp8,
-  kH264,
-};
+enum VideoCodec { kVp8, kH264, };
+
+enum AudioCodec { kOpus, kPcm16, kExternalAudio, };
 
-enum AudioCodec {
-  kOpus,
-  kPcm16,
-  kExternalAudio,
+struct RtpConfig {
+  RtpConfig();
+  int history_ms;  // The time RTP packets are stored for retransmissions.
+  int max_delay_ms;
+  int payload_type;
 };
 
 struct CastTransportConfig {
@@ -38,10 +39,8 @@ struct CastTransportConfig {
   ~CastTransportConfig();
 
   // Transport: Local receiver.
-  std::string receiver_ip_address;
-  std::string local_ip_address;
-  int receive_port;
-  int send_port;
+  net::IPEndPoint receiver_endpoint;
+  net::IPEndPoint local_endpoint;
 
   uint32 audio_ssrc;
   uint32 video_ssrc;
@@ -50,17 +49,13 @@ struct CastTransportConfig {
   AudioCodec audio_codec;
 
   // RTP.
-  int audio_rtp_history_ms;
-  int video_rtp_history_ms;
-  int audio_rtp_max_delay_ms;
-  int video_rtp_max_delay_ms;
-  int audio_rtp_payload_type;
-  int video_rtp_payload_type;
+  RtpConfig audio_rtp_config;
+  RtpConfig video_rtp_config;
 
   int audio_frequency;
   int audio_channels;
 
-  std::string aes_key;  // Binary string of size kAesKeySize.
+  std::string aes_key;      // Binary string of size kAesKeySize.
   std::string aes_iv_mask;  // Binary string of size kAesBlockSize.
 };
 
@@ -72,6 +67,7 @@ struct EncodedVideoFrame {
   bool key_frame;
   uint32 frame_id;
   uint32 last_referenced_frame_id;
+  uint32 rtp_timestamp;
   std::string data;
 };
 
@@ -81,8 +77,7 @@ struct EncodedAudioFrame {
 
   AudioCodec codec;
   uint32 frame_id;  // Needed to release the frame.
-  int samples;  // Needed send side to advance the RTP timestamp.
-                // Not used receive side.
+  uint32 rtp_timestamp;
   // Support for max sampling rate of 48KHz, 2 channels, 100 ms duration.
   static const int kMaxNumberOfSamples = 48 * 2 * 100;
   std::string data;
@@ -91,21 +86,7 @@ struct EncodedAudioFrame {
 typedef std::vector<uint8> Packet;
 typedef std::vector<Packet> PacketList;
 
-class PacketReceiver : public base::RefCountedThreadSafe<PacketReceiver> {
- public:
-  // All packets received from the network should be delivered via this
-  // function.
-  virtual void ReceivedPacket(const uint8* packet, size_t length,
-                              const base::Closure callback) = 0;
-
-  static void DeletePacket(const uint8* packet);
-
- protected:
-  virtual ~PacketReceiver() {}
-
- private:
-  friend class base::RefCountedThreadSafe<PacketReceiver>;
-};
+typedef base::Callback<void(scoped_ptr<Packet> packet)> PacketReceiverCallback;
 
 class PacketSender {
  public:
@@ -132,7 +113,7 @@ struct RtcpSenderFrameLogMessage {
   uint32 rtp_timestamp;
 };
 
-typedef std::list<RtcpSenderFrameLogMessage> RtcpSenderLogMessage;
+typedef std::vector<RtcpSenderFrameLogMessage> RtcpSenderLogMessage;
 
 struct RtcpSenderInfo {
   RtcpSenderInfo();
@@ -150,7 +131,7 @@ struct RtcpReportBlock {
   RtcpReportBlock();
   ~RtcpReportBlock();
   uint32 remote_ssrc;  // SSRC of sender of this report.
-  uint32 media_ssrc;  // SSRC of the RTP packet sender.
+  uint32 media_ssrc;   // SSRC of the RTP packet sender.
   uint8 fraction_lost;
   uint32 cumulative_lost;  // 24 bits valid.
   uint32 extended_high_sequence_number;
@@ -166,16 +147,26 @@ struct RtcpDlrrReportBlock {
   uint32 delay_since_last_rr;
 };
 
+// This is only needed because IPC messages don't support more than
+// 5 arguments.
+struct SendRtcpFromRtpSenderData {
+  SendRtcpFromRtpSenderData();
+  ~SendRtcpFromRtpSenderData();
+  uint32 packet_type_flags;
+  uint32 sending_ssrc;
+  std::string c_name;
+};
+
 inline bool operator==(RtcpSenderInfo lhs, RtcpSenderInfo rhs) {
   return lhs.ntp_seconds == rhs.ntp_seconds &&
-      lhs.ntp_fraction == rhs.ntp_fraction &&
-      lhs.rtp_timestamp == rhs.rtp_timestamp &&
-      lhs.send_packet_count == rhs.send_packet_count &&
-      lhs.send_octet_count == rhs.send_octet_count;
+         lhs.ntp_fraction == rhs.ntp_fraction &&
+         lhs.rtp_timestamp == rhs.rtp_timestamp &&
+         lhs.send_packet_count == rhs.send_packet_count &&
+         lhs.send_octet_count == rhs.send_octet_count;
 }
 
 }  // namespace transport
 }  // namespace cast
 }  // namespace media
 
-#endif // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_CONFIG_H_
+#endif  // MEDIA_CAST_TRANSPORT_CAST_TRANSPORT_CONFIG_H_