Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -22,27 +22,28 @@ namespace webrtc {
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class CriticalSectionWrapper;
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class ModuleRtpRtcpImpl;
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class ReceiverFEC;
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class RTPReceiver;
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class RTPPayloadRegistry;
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class RtpReceiver;
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class RTPReceiverVideo : public RTPReceiverStrategy {
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public:
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RTPReceiverVideo(const int32_t id,
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const RTPPayloadRegistry* rtp_payload_registry,
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RtpData* data_callback);
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RTPReceiverVideo(const int32_t id, RtpData* data_callback);
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virtual ~RTPReceiverVideo();
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virtual int32_t ParseRtpPacket(
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WebRtcRTPHeader* rtp_header,
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const ModuleRTPUtility::PayloadUnion& specific_payload,
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const bool is_red,
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const PayloadUnion& specific_payload,
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bool is_red,
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const uint8_t* packet,
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const uint16_t packet_length,
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const int64_t timestamp,
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const bool is_first_packet) OVERRIDE;
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uint16_t packet_length,
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int64_t timestamp,
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bool is_first_packet) OVERRIDE;
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virtual int32_t GetFrequencyHz() const OVERRIDE;
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TelephoneEventHandler* GetTelephoneEventHandler() {
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return NULL;
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}
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int GetPayloadTypeFrequency() const OVERRIDE;
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virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const
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OVERRIDE;
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@ -51,41 +52,32 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
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virtual int32_t OnNewPayloadTypeCreated(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type,
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const uint32_t frequency) OVERRIDE;
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int8_t payload_type,
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uint32_t frequency) OVERRIDE;
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virtual int32_t InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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const int32_t id,
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const int8_t payload_type,
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int32_t id,
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int8_t payload_type,
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const ModuleRTPUtility::PayloadUnion& specific_payload) const OVERRIDE;
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virtual int32_t ReceiveRecoveredPacketCallback(
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WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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const uint16_t payload_data_length);
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const PayloadUnion& specific_payload) const OVERRIDE;
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void SetPacketOverHead(uint16_t packet_over_head);
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protected:
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int32_t SetCodecType(const RtpVideoCodecTypes video_type,
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WebRtcRTPHeader* rtp_header) const;
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int32_t ParseVideoCodecSpecificSwitch(
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WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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const uint16_t payload_data_length,
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const RtpVideoCodecTypes video_type,
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const bool is_first_packet);
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uint16_t payload_data_length,
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bool is_first_packet);
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int32_t ReceiveGenericCodec(WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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const uint16_t payload_data_length);
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uint16_t payload_data_length);
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int32_t ReceiveVp8Codec(WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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const uint16_t payload_data_length);
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uint16_t payload_data_length);
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int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
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uint8_t* data_buffer) const;
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@ -94,21 +86,17 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
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int32_t ParseVideoCodecSpecific(
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WebRtcRTPHeader* rtp_header,
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const uint8_t* payload_data,
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const uint16_t payload_data_length,
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const RtpVideoCodecTypes video_type,
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const bool is_red,
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uint16_t payload_data_length,
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RtpVideoCodecTypes video_type,
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bool is_red,
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const uint8_t* incoming_rtp_packet,
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const uint16_t incoming_rtp_packet_size,
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const int64_t now_ms,
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const bool is_first_packet);
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uint16_t incoming_rtp_packet_size,
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int64_t now_ms,
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bool is_first_packet);
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int32_t id_;
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const RTPPayloadRegistry* rtp_rtp_payload_registry_;
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CriticalSectionWrapper* critical_section_receiver_video_;
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// FEC
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bool current_fec_frame_decoded_;
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ReceiverFEC* receive_fec_;
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};
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} // namespace webrtc
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