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Revert r4301

Browse Source 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tnakamura@webrtc.org
2013-07-16 19:25:04 +00:00
parent 7b2f955e56
commit aa4d96a134
98 changed files with 4306 additions and 2856 deletions
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118
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -17,6 +17,8 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
@ -41,12 +43,72 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Receiver part.
// Configure a timeout value.
virtual int32_t SetPacketTimeout(const uint32_t rtp_timeout_ms,
const uint32_t rtcp_timeout_ms);
// Set periodic dead or alive notification.
virtual int32_t SetPeriodicDeadOrAliveStatus(
const bool enable,
const uint8_t sample_time_seconds);
// Get periodic dead or alive notification status.
virtual int32_t PeriodicDeadOrAliveStatus(
bool& enable,
uint8_t& sample_time_seconds);
virtual int32_t RegisterReceivePayload(const CodecInst& voice_codec);
virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec);
virtual int32_t ReceivePayloadType(const CodecInst& voice_codec,
int8_t* pl_type);
virtual int32_t ReceivePayloadType(const VideoCodec& video_codec,
int8_t* pl_type);
virtual int32_t DeRegisterReceivePayload(
const int8_t payload_type);
// Get the currently configured SSRC filter.
virtual int32_t SSRCFilter(uint32_t& allowed_ssrc) const;
// Set a SSRC to be used as a filter for incoming RTP streams.
virtual int32_t SetSSRCFilter(const bool enable,
const uint32_t allowed_ssrc);
// Get last received remote timestamp.
virtual uint32_t RemoteTimestamp() const;
// Get the local time of the last received remote timestamp.
virtual int64_t LocalTimeOfRemoteTimeStamp() const;
// Get the current estimated remote timestamp.
virtual int32_t EstimatedRemoteTimeStamp(
uint32_t& timestamp) const;
virtual uint32_t RemoteSSRC() const;
virtual int32_t RemoteCSRCs(
uint32_t arr_of_csrc[kRtpCsrcSize]) const;
virtual int32_t SetRTXReceiveStatus(const bool enable,
const uint32_t ssrc);
virtual int32_t RTXReceiveStatus(bool* enable, uint32_t* ssrc,
int* payloadType) const;
virtual void SetRtxReceivePayloadType(int payload_type);
// Called when we receive an RTP packet.
virtual int32_t IncomingRtpPacket(const uint8_t* incoming_packet,
const uint16_t packet_length,
const RTPHeader& parsed_rtp_header);
// Called when we receive an RTCP packet.
virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
uint16_t incoming_packet_length);
virtual void SetRemoteSSRC(const uint32_t ssrc);
// Sender part.
virtual int32_t RegisterSendPayload(const CodecInst& voice_codec);
@ -177,11 +239,32 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Normal SR and RR are triggered via the process function.
virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport);
// Statistics of our locally created statistics of the received RTP stream.
virtual int32_t StatisticsRTP(uint8_t* fraction_lost,
uint32_t* cum_lost,
uint32_t* ext_max,
uint32_t* jitter,
uint32_t* max_jitter = NULL) const;
// Reset RTP statistics.
virtual int32_t ResetStatisticsRTP();
virtual int32_t ResetReceiveDataCountersRTP();
virtual int32_t ResetSendDataCountersRTP();
// Statistics of the amount of data sent and received.
virtual int32_t DataCountersRTP(uint32_t* bytes_sent,
uint32_t* packets_sent) const;
uint32_t* packets_sent,
uint32_t* bytes_received,
uint32_t* packets_received) const;
virtual int32_t ReportBlockStatistics(
uint8_t* fraction_lost,
uint32_t* cum_lost,
uint32_t* ext_max,
uint32_t* jitter,
uint32_t* jitter_transmission_time_offset);
// Get received RTCP report, sender info.
virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info);
@ -230,6 +313,13 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// (NACK) Negative acknowledgment part.
// Is Negative acknowledgment requests on/off?
virtual NACKMethod NACK() const;
// Turn negative acknowledgment requests on/off.
virtual int32_t SetNACKStatus(const NACKMethod method,
int max_reordering_threshold);
virtual int SelectiveRetransmissions() const;
virtual int SetSelectiveRetransmissions(uint8_t settings);
@ -259,6 +349,12 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
virtual int32_t SetAudioPacketSize(
const uint16_t packet_size_samples);
// Forward DTMFs to decoder for playout.
virtual int SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
// Is forwarding of outband telephone events turned on/off?
virtual bool TelephoneEventForwardToDecoder() const;
virtual bool SendTelephoneEventActive(int8_t& telephone_event) const;
// Send a TelephoneEvent tone using RFC 2833 (4733).
@ -286,6 +382,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Video part.
virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
virtual RtpVideoCodecTypes SendVideoCodec() const;
virtual int32_t SendRTCPSliceLossIndication(
@ -327,6 +425,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
uint32_t* fec_rate,
uint32_t* nackRate) const;
virtual void SetRemoteSSRC(const uint32_t ssrc);
virtual uint32_t SendTimeOfSendReport(const uint32_t send_report);
// Good state of RTP receiver inform sender.
@ -356,6 +456,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
bool UpdateRTCPReceiveInformationTimers();
void ProcessDeadOrAliveTimer();
uint32_t BitrateReceivedNow() const;
// Get remote SequenceNumber.
@ -364,7 +466,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Only for internal testing.
uint32_t LastSendReport(uint32_t& last_rtcptime);
RTPPayloadRegistry rtp_payload_registry_;
RTPSender rtp_sender_;
scoped_ptr<RTPReceiver> rtp_receiver_;
RTCPSender rtcp_sender_;
RTCPReceiver rtcp_receiver_;
@ -374,13 +479,14 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
private:
int64_t RtcpReportInterval();
ReceiveStatistics* receive_statistics_;
RTPReceiverAudio* rtp_telephone_event_handler_;
int32_t id_;
const bool audio_;
bool collision_detected_;
int64_t last_process_time_;
int64_t last_bitrate_process_time_;
int64_t last_packet_timeout_process_time_;
int64_t last_rtt_process_time_;
uint16_t packet_overhead_;
@ -389,6 +495,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
ModuleRtpRtcpImpl* default_module_;
std::list<ModuleRtpRtcpImpl*> child_modules_;
// Dead or alive.
bool dead_or_alive_active_;
uint32_t dead_or_alive_timeout_ms_;
int64_t dead_or_alive_last_timer_;
// Send side
NACKMethod nack_method_;
uint32_t nack_last_time_sent_full_;