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
This commit is contained in:
wu@webrtc.org
@ -17,8 +17,6 @@
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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@ -43,72 +41,12 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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// Receiver part.
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// Configure a timeout value.
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virtual int32_t SetPacketTimeout(const uint32_t rtp_timeout_ms,
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const uint32_t rtcp_timeout_ms) OVERRIDE;
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// Set periodic dead or alive notification.
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virtual int32_t SetPeriodicDeadOrAliveStatus(
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const bool enable,
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const uint8_t sample_time_seconds) OVERRIDE;
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// Get periodic dead or alive notification status.
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virtual int32_t PeriodicDeadOrAliveStatus(
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bool& enable,
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uint8_t& sample_time_seconds) OVERRIDE;
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virtual int32_t RegisterReceivePayload(const CodecInst& voice_codec) OVERRIDE;
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virtual int32_t RegisterReceivePayload(
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const VideoCodec& video_codec) OVERRIDE;
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virtual int32_t ReceivePayloadType(const CodecInst& voice_codec,
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int8_t* pl_type) OVERRIDE;
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virtual int32_t ReceivePayloadType(const VideoCodec& video_codec,
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int8_t* pl_type) OVERRIDE;
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virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) OVERRIDE;
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// Get the currently configured SSRC filter.
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virtual int32_t SSRCFilter(uint32_t& allowed_ssrc) const OVERRIDE;
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// Set a SSRC to be used as a filter for incoming RTP streams.
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virtual int32_t SetSSRCFilter(const bool enable,
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const uint32_t allowed_ssrc) OVERRIDE;
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// Get last received remote timestamp.
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virtual uint32_t RemoteTimestamp() const OVERRIDE;
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// Get the local time of the last received remote timestamp.
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virtual int64_t LocalTimeOfRemoteTimeStamp() const OVERRIDE;
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// Get the current estimated remote timestamp.
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virtual int32_t EstimatedRemoteTimeStamp(uint32_t& timestamp) const OVERRIDE;
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virtual uint32_t RemoteSSRC() const OVERRIDE;
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virtual int32_t RemoteCSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const
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OVERRIDE;
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virtual int32_t SetRTXReceiveStatus(const bool enable,
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const uint32_t ssrc) OVERRIDE;
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virtual int32_t RTXReceiveStatus(bool* enable, uint32_t* ssrc,
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int* payloadType) const OVERRIDE;
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virtual void SetRtxReceivePayloadType(int payload_type) OVERRIDE;
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// Called when we receive an RTP packet.
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virtual int32_t IncomingRtpPacket(
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const uint8_t* incoming_packet,
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const uint16_t packet_length,
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const RTPHeader& parsed_rtp_header) OVERRIDE;
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// Called when we receive an RTCP packet.
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virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
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uint16_t incoming_packet_length) OVERRIDE;
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virtual void SetRemoteSSRC(const uint32_t ssrc);
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// Sender part.
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virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) OVERRIDE;
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@ -239,32 +177,11 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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// Normal SR and RR are triggered via the process function.
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virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport) OVERRIDE;
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// Statistics of our locally created statistics of the received RTP stream.
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virtual int32_t StatisticsRTP(uint8_t* fraction_lost,
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uint32_t* cum_lost,
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uint32_t* ext_max,
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uint32_t* jitter,
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uint32_t* max_jitter = NULL) const OVERRIDE;
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// Reset RTP statistics.
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virtual int32_t ResetStatisticsRTP() OVERRIDE;
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virtual int32_t ResetReceiveDataCountersRTP() OVERRIDE;
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virtual int32_t ResetSendDataCountersRTP() OVERRIDE;
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// Statistics of the amount of data sent and received.
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virtual int32_t DataCountersRTP(uint32_t* bytes_sent,
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uint32_t* packets_sent,
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uint32_t* bytes_received,
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uint32_t* packets_received) const OVERRIDE;
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virtual int32_t ReportBlockStatistics(
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uint8_t* fraction_lost,
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uint32_t* cum_lost,
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uint32_t* ext_max,
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uint32_t* jitter,
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uint32_t* jitter_transmission_time_offset);
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uint32_t* packets_sent) const OVERRIDE;
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// Get received RTCP report, sender info.
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virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) OVERRIDE;
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@ -313,13 +230,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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// (NACK) Negative acknowledgment part.
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// Is Negative acknowledgment requests on/off?
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virtual NACKMethod NACK() const OVERRIDE;
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// Turn negative acknowledgment requests on/off.
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virtual int32_t SetNACKStatus(const NACKMethod method,
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int max_reordering_threshold) OVERRIDE;
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virtual int SelectiveRetransmissions() const OVERRIDE;
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virtual int SetSelectiveRetransmissions(uint8_t settings) OVERRIDE;
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@ -333,6 +243,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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virtual int32_t SetStorePacketsStatus(
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const bool enable, const uint16_t number_to_store) OVERRIDE;
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virtual bool StorePackets() const OVERRIDE;
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// (APP) Application specific data.
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virtual int32_t SetRTCPApplicationSpecificData(
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const uint8_t sub_type,
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@ -350,13 +262,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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virtual int32_t SetAudioPacketSize(
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const uint16_t packet_size_samples) OVERRIDE;
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// Forward DTMFs to decoder for playout.
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virtual int SetTelephoneEventForwardToDecoder(
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bool forward_to_decoder) OVERRIDE;
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// Is forwarding of outband telephone events turned on/off?
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virtual bool TelephoneEventForwardToDecoder() const OVERRIDE;
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virtual bool SendTelephoneEventActive(int8_t& telephone_event) const OVERRIDE;
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// Send a TelephoneEvent tone using RFC 2833 (4733).
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@ -384,8 +289,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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// Video part.
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virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
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virtual RtpVideoCodecTypes SendVideoCodec() const;
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virtual int32_t SendRTCPSliceLossIndication(
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@ -427,8 +330,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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uint32_t* fec_rate,
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uint32_t* nackRate) const OVERRIDE;
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virtual void SetRemoteSSRC(const uint32_t ssrc);
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virtual uint32_t SendTimeOfSendReport(const uint32_t send_report);
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// Good state of RTP receiver inform sender.
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@ -458,8 +359,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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bool UpdateRTCPReceiveInformationTimers();
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void ProcessDeadOrAliveTimer();
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uint32_t BitrateReceivedNow() const;
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// Get remote SequenceNumber.
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@ -468,10 +367,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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// Only for internal testing.
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uint32_t LastSendReport(uint32_t& last_rtcptime);
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RTPPayloadRegistry rtp_payload_registry_;
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RTPSender rtp_sender_;
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scoped_ptr<RTPReceiver> rtp_receiver_;
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RTCPSender rtcp_sender_;
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RTCPReceiver rtcp_receiver_;
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@ -481,14 +377,13 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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private:
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int64_t RtcpReportInterval();
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RTPReceiverAudio* rtp_telephone_event_handler_;
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ReceiveStatistics* receive_statistics_;
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int32_t id_;
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const bool audio_;
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bool collision_detected_;
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int64_t last_process_time_;
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int64_t last_bitrate_process_time_;
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int64_t last_packet_timeout_process_time_;
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int64_t last_rtt_process_time_;
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uint16_t packet_overhead_;
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@ -497,10 +392,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
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ModuleRtpRtcpImpl* default_module_;
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std::list<ModuleRtpRtcpImpl*> child_modules_;
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// Dead or alive.
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bool dead_or_alive_active_;
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uint32_t dead_or_alive_timeout_ms_;
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int64_t dead_or_alive_last_timer_;
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// Send side
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NACKMethod nack_method_;
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uint32_t nack_last_time_sent_full_;
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