pre-factor cleanup pre-work.
Review URL: https://webrtc-codereview.appspot.com/938010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
parent
4cebe6cded
commit
c66e8b3f31
@ -52,7 +52,11 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
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}
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ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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: _rtpSender(configuration.id, configuration.audio, configuration.clock),
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: _rtpSender(configuration.id,
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configuration.audio,
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configuration.clock,
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configuration.outgoing_transport,
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configuration.audio_messages),
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_rtpReceiver(configuration.id, configuration.audio, configuration.clock,
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this),
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_rtcpSender(configuration.id, configuration.audio, configuration.clock,
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@ -97,10 +101,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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_rtcpReceiver.RegisterRtcpObservers(configuration.intra_frame_callback,
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configuration.bandwidth_callback,
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configuration.rtcp_feedback);
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_rtpSender.RegisterAudioCallback(configuration.audio_messages);
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_rtpReceiver.RegisterIncomingAudioCallback(configuration.audio_messages);
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_rtpSender.RegisterSendTransport(configuration.outgoing_transport);
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_rtcpSender.RegisterSendTransport(configuration.outgoing_transport);
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// make sure that RTCP objects are aware of our SSRC
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@ -628,7 +630,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetStartTimestamp(
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"SetStartTimestamp(%d)",
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timestamp);
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_rtcpSender.SetStartTimestamp(timestamp);
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return _rtpSender.SetStartTimestamp(timestamp, true);
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_rtpSender.SetStartTimestamp(timestamp, true);
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return 0; // TODO(pwestin): change to void.
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}
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WebRtc_UWord16 ModuleRtpRtcpImpl::SequenceNumber() const {
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@ -646,7 +649,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetSequenceNumber(
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"SetSequenceNumber(%d)",
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seqNum);
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return _rtpSender.SetSequenceNumber(seqNum);
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_rtpSender.SetSequenceNumber(seqNum);
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return 0; // TODO(pwestin): change to void.
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}
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WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
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@ -659,17 +663,16 @@ WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
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WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRC(const WebRtc_UWord32 ssrc) {
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WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetSSRC(%d)", ssrc);
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if (_rtpSender.SetSSRC(ssrc) == 0) {
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_rtcpReceiver.SetSSRC(ssrc);
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_rtcpSender.SetSSRC(ssrc);
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return 0;
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}
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return -1;
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_rtpSender.SetSSRC(ssrc);
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_rtcpReceiver.SetSSRC(ssrc);
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_rtcpSender.SetSSRC(ssrc);
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return 0; // TODO(pwestin): change to void.
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}
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WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) {
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_rtcpSender.SetCSRCStatus(include);
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return _rtpSender.SetCSRCStatus(include);
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_rtpSender.SetCSRCStatus(include);
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return 0; // TODO(pwestin): change to void.
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}
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WebRtc_Word32 ModuleRtpRtcpImpl::CSRCs(
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@ -702,16 +705,15 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCs(
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}
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it++;
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}
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return 0;
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} else {
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for (int i = 0; i < arrLength; i++) {
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WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "\tidx:%d CSRC:%u", i,
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arrOfCSRC[i]);
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}
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_rtcpSender.SetCSRCs(arrOfCSRC, arrLength);
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return _rtpSender.SetCSRCs(arrOfCSRC, arrLength);
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_rtpSender.SetCSRCs(arrOfCSRC, arrLength);
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}
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return 0; // TODO(pwestin): change to void.
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}
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WebRtc_UWord32 ModuleRtpRtcpImpl::PacketCountSent() const {
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@ -1129,7 +1131,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
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WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
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"ResetSendDataCountersRTP()");
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return _rtpSender.ResetDataCounters();
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_rtpSender.ResetDataCounters();
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return 0; // TODO(pwestin): change to void.
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}
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// Force a send of an RTCP packet
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@ -1495,7 +1498,8 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SetStorePacketsStatus(
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WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
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"SetStorePacketsStatus(disable)");
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}
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return _rtpSender.SetStorePacketsStatus(enable, numberToStore);
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_rtpSender.SetStorePacketsStatus(enable, numberToStore);
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return 0; // TODO(pwestin): change to void.
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}
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/*
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File diff suppressed because it is too large
Load Diff
@ -15,9 +15,9 @@
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#include <cmath>
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#include <map>
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#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
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#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
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#include "rtp_rtcp_defines.h"
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#include "common_types.h" // Encryption
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#include "common_types.h"
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#include "ssrc_database.h"
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#include "Bitrate.h"
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#include "rtp_header_extension.h"
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@ -32,323 +32,315 @@ class RTPPacketHistory;
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class RTPSenderAudio;
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class RTPSenderVideo;
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class RTPSenderInterface
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{
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public:
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RTPSenderInterface() {}
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virtual ~RTPSenderInterface() {}
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class RTPSenderInterface {
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public:
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RTPSenderInterface() {}
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virtual ~RTPSenderInterface() {}
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virtual WebRtc_UWord32 SSRC() const = 0;
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virtual WebRtc_UWord32 Timestamp() const = 0;
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virtual WebRtc_UWord32 SSRC() const = 0;
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virtual WebRtc_UWord32 Timestamp() const = 0;
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virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
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virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
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const WebRtc_Word8 payloadType,
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const bool markerBit,
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const WebRtc_UWord32 captureTimeStamp,
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const bool timeStampProvided = true,
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const bool incSequenceNumber = true) = 0;
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virtual WebRtc_UWord16 RTPHeaderLength() const = 0;
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virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
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virtual WebRtc_UWord16 SequenceNumber() const = 0;
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virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
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virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
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virtual WebRtc_UWord16 PacketOverHead() const = 0;
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virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
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virtual WebRtc_UWord16 RTPHeaderLength() const = 0;
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virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
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virtual WebRtc_UWord16 SequenceNumber() const = 0;
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virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
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virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
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virtual WebRtc_UWord16 PacketOverHead() const = 0;
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virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
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virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* data_buffer,
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const WebRtc_UWord16 payload_length,
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const WebRtc_UWord16 rtp_header_length,
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int64_t capture_time_ms,
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const StorageType storage) = 0;
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virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* data_buffer,
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WebRtc_UWord16 payload_length,
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WebRtc_UWord16 rtp_header_length,
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int64_t capture_time_ms,
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StorageType storage) = 0;
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};
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class RTPSender : public Bitrate, public RTPSenderInterface
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{
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public:
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RTPSender(const WebRtc_Word32 id, const bool audio, RtpRtcpClock* clock);
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virtual ~RTPSender();
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class RTPSender : public Bitrate, public RTPSenderInterface {
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public:
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RTPSender(const WebRtc_Word32 id, const bool audio, RtpRtcpClock* clock,
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Transport* transport, RtpAudioFeedback* audio_feedback);
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virtual ~RTPSender();
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void ProcessBitrate();
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void ProcessSendToNetwork();
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void ProcessBitrate();
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void ProcessSendToNetwork();
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WebRtc_UWord16 ActualSendBitrateKbit() const;
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WebRtc_UWord16 ActualSendBitrateKbit() const;
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WebRtc_UWord32 VideoBitrateSent() const;
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WebRtc_UWord32 FecOverheadRate() const;
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WebRtc_UWord32 NackOverheadRate() const;
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WebRtc_UWord32 VideoBitrateSent() const;
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WebRtc_UWord32 FecOverheadRate() const;
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WebRtc_UWord32 NackOverheadRate() const;
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void SetTargetSendBitrate(const WebRtc_UWord32 bits);
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void SetTargetSendBitrate(const WebRtc_UWord32 bits);
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WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers
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WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers
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// callback
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WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
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WebRtc_Word32 RegisterPayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate);
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WebRtc_Word32 RegisterPayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate);
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WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
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WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
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WebRtc_Word8 SendPayloadType() const;
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WebRtc_Word8 SendPayloadType() const;
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int SendPayloadFrequency() const;
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int SendPayloadFrequency() const;
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void SetSendingStatus(const bool enabled);
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void SetSendingStatus(const bool enabled);
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void SetSendingMediaStatus(const bool enabled);
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bool SendingMedia() const;
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void SetSendingMediaStatus(const bool enabled);
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bool SendingMedia() const;
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// number of sent RTP packets
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WebRtc_UWord32 Packets() const;
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// number of sent RTP packets
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WebRtc_UWord32 Packets() const;
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// number of sent RTP bytes
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WebRtc_UWord32 Bytes() const;
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// number of sent RTP bytes
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WebRtc_UWord32 Bytes() const;
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void ResetDataCounters();
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WebRtc_Word32 ResetDataCounters();
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WebRtc_UWord32 StartTimestamp() const;
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void SetStartTimestamp(WebRtc_UWord32 timestamp, bool force);
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WebRtc_UWord32 StartTimestamp() const;
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WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp,
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const bool force = false);
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WebRtc_UWord32 GenerateNewSSRC();
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void SetSSRC(const WebRtc_UWord32 ssrc);
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WebRtc_UWord32 GenerateNewSSRC();
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WebRtc_Word32 SetSSRC( const WebRtc_UWord32 ssrc);
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WebRtc_UWord16 SequenceNumber() const;
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void SetSequenceNumber(WebRtc_UWord16 seq);
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WebRtc_UWord16 SequenceNumber() const;
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WebRtc_Word32 SetSequenceNumber( WebRtc_UWord16 seq);
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WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
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WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
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void SetCSRCStatus(const bool include);
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WebRtc_Word32 SetCSRCStatus(const bool include);
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void SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength);
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WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength);
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WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
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const WebRtc_UWord16 packetOverHead);
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WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
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const WebRtc_UWord16 packetOverHead);
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WebRtc_Word32 SendOutgoingData(const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 timeStamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation,
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VideoCodecInformation* codecInfo = NULL,
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const RTPVideoTypeHeader* rtpTypeHdr = NULL);
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WebRtc_Word32 SendOutgoingData(const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 timeStamp,
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int64_t capture_time_ms,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 payloadSize,
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const RTPFragmentationHeader* fragmentation,
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VideoCodecInformation* codecInfo = NULL,
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const RTPVideoTypeHeader* rtpTypeHdr = NULL);
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WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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int64_t capture_time_ms,
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WebRtc_Word32 bytes);
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/*
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* RTP header extension
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*/
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WebRtc_Word32 SetTransmissionTimeOffset(
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const WebRtc_Word32 transmissionTimeOffset);
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WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
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WebRtc_UWord32 capture_timestamp,
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int64_t capture_time_ms,
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WebRtc_Word32 bytes);
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/*
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* RTP header extension
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*/
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WebRtc_Word32 SetTransmissionTimeOffset(
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const WebRtc_Word32 transmissionTimeOffset);
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WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
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const WebRtc_UWord8 id);
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WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
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const WebRtc_UWord8 id);
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WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
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WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
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WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
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WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
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WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const;
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WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const;
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WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
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WebRtc_UWord8* dataBuffer) const;
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WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
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WebRtc_UWord8* dataBuffer) const;
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bool UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
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const WebRtc_UWord16 rtp_packet_length,
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const WebRtcRTPHeader& rtp_header,
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const WebRtc_Word64 time_diff_ms) const;
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bool UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
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const WebRtc_UWord16 rtp_packet_length,
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const WebRtcRTPHeader& rtp_header,
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const WebRtc_Word64 time_diff_ms) const;
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void SetTransmissionSmoothingStatus(const bool enable);
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void SetTransmissionSmoothingStatus(const bool enable);
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bool TransmissionSmoothingStatus() const;
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bool TransmissionSmoothingStatus() const;
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/*
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/*
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* NACK
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*/
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int SelectiveRetransmissions() const;
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int SetSelectiveRetransmissions(uint8_t settings);
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void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
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const WebRtc_UWord16* nackSequenceNumbers,
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const WebRtc_UWord16 avgRTT);
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int SelectiveRetransmissions() const;
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int SetSelectiveRetransmissions(uint8_t settings);
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void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
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const WebRtc_UWord16* nackSequenceNumbers,
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const WebRtc_UWord16 avgRTT);
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WebRtc_Word32 SetStorePacketsStatus(const bool enable,
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const WebRtc_UWord16 numberToStore);
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void SetStorePacketsStatus(const bool enable,
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const WebRtc_UWord16 numberToStore);
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bool StorePackets() const;
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bool StorePackets() const;
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WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
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WebRtc_UWord32 min_resend_time = 0);
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WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
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WebRtc_UWord32 min_resend_time = 0);
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WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8* packet,
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const WebRtc_UWord32 size);
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WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8* packet,
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const WebRtc_UWord32 size);
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bool ProcessNACKBitRate(const WebRtc_UWord32 now);
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bool ProcessNACKBitRate(const WebRtc_UWord32 now);
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/*
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/*
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* RTX
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*/
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void SetRTXStatus(const bool enable,
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const bool setSSRC,
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const WebRtc_UWord32 SSRC);
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void SetRTXStatus(const bool enable,
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const bool setSSRC,
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const WebRtc_UWord32 SSRC);
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void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
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void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
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/*
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/*
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* Functions wrapping RTPSenderInterface
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*/
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virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
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const WebRtc_Word8 payloadType,
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const bool markerBit,
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const WebRtc_UWord32 captureTimeStamp,
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const bool timeStampProvided = true,
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const bool incSequenceNumber = true);
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virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
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||||
const WebRtc_Word8 payloadType,
|
||||
const bool markerBit,
|
||||
const WebRtc_UWord32 captureTimeStamp,
|
||||
const bool timeStampProvided = true,
|
||||
const bool incSequenceNumber = true);
|
||||
|
||||
virtual WebRtc_UWord16 RTPHeaderLength() const ;
|
||||
virtual WebRtc_UWord16 IncrementSequenceNumber();
|
||||
virtual WebRtc_UWord16 MaxPayloadLength() const;
|
||||
virtual WebRtc_UWord16 PacketOverHead() const;
|
||||
virtual WebRtc_UWord16 RTPHeaderLength() const ;
|
||||
virtual WebRtc_UWord16 IncrementSequenceNumber();
|
||||
virtual WebRtc_UWord16 MaxPayloadLength() const;
|
||||
virtual WebRtc_UWord16 PacketOverHead() const;
|
||||
|
||||
// current timestamp
|
||||
virtual WebRtc_UWord32 Timestamp() const;
|
||||
virtual WebRtc_UWord32 SSRC() const;
|
||||
// current timestamp
|
||||
virtual WebRtc_UWord32 Timestamp() const;
|
||||
virtual WebRtc_UWord32 SSRC() const;
|
||||
|
||||
virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* data_buffer,
|
||||
const WebRtc_UWord16 payload_length,
|
||||
const WebRtc_UWord16 rtp_header_length,
|
||||
int64_t capture_time_ms,
|
||||
const StorageType storage);
|
||||
|
||||
/*
|
||||
virtual WebRtc_Word32 SendToNetwork(WebRtc_UWord8* data_buffer,
|
||||
WebRtc_UWord16 payload_length,
|
||||
WebRtc_UWord16 rtp_header_length,
|
||||
int64_t capture_time_ms,
|
||||
StorageType storage);
|
||||
/*
|
||||
* Audio
|
||||
*/
|
||||
WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
|
||||
// Send a DTMF tone using RFC 2833 (4733)
|
||||
WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
|
||||
const WebRtc_UWord16 time_ms,
|
||||
const WebRtc_UWord8 level);
|
||||
|
||||
// Send a DTMF tone using RFC 2833 (4733)
|
||||
WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
|
||||
const WebRtc_UWord16 time_ms,
|
||||
const WebRtc_UWord8 level);
|
||||
bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
|
||||
|
||||
bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
|
||||
// Set audio packet size, used to determine when it's time to send a DTMF
|
||||
// packet in silence (CNG)
|
||||
WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
|
||||
|
||||
// set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
|
||||
WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
|
||||
// Set status and ID for header-extension-for-audio-level-indication.
|
||||
WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
|
||||
const WebRtc_UWord8 ID);
|
||||
|
||||
// Set status and ID for header-extension-for-audio-level-indication.
|
||||
WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
|
||||
const WebRtc_UWord8 ID);
|
||||
// Get status and ID for header-extension-for-audio-level-indication.
|
||||
WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
|
||||
WebRtc_UWord8& ID) const;
|
||||
|
||||
// Get status and ID for header-extension-for-audio-level-indication.
|
||||
WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
|
||||
WebRtc_UWord8& ID) const;
|
||||
// Store the audio level in dBov for
|
||||
// header-extension-for-audio-level-indication.
|
||||
WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
|
||||
|
||||
// Store the audio level in dBov for header-extension-for-audio-level-indication.
|
||||
WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
|
||||
// Set payload type for Redundant Audio Data RFC 2198
|
||||
WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
|
||||
|
||||
// Set payload type for Redundant Audio Data RFC 2198
|
||||
WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
|
||||
// Get payload type for Redundant Audio Data RFC 2198
|
||||
WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
|
||||
|
||||
// Get payload type for Redundant Audio Data RFC 2198
|
||||
WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
|
||||
|
||||
/*
|
||||
/*
|
||||
* Video
|
||||
*/
|
||||
VideoCodecInformation* CodecInformationVideo();
|
||||
VideoCodecInformation* CodecInformationVideo();
|
||||
|
||||
RtpVideoCodecTypes VideoCodecType() const;
|
||||
RtpVideoCodecTypes VideoCodecType() const;
|
||||
|
||||
WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
|
||||
WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
|
||||
|
||||
WebRtc_Word32 SendRTPIntraRequest();
|
||||
WebRtc_Word32 SendRTPIntraRequest();
|
||||
|
||||
// FEC
|
||||
WebRtc_Word32 SetGenericFECStatus(const bool enable,
|
||||
const WebRtc_UWord8 payloadTypeRED,
|
||||
const WebRtc_UWord8 payloadTypeFEC);
|
||||
// FEC
|
||||
WebRtc_Word32 SetGenericFECStatus(const bool enable,
|
||||
const WebRtc_UWord8 payloadTypeRED,
|
||||
const WebRtc_UWord8 payloadTypeFEC);
|
||||
|
||||
WebRtc_Word32 GenericFECStatus(bool& enable,
|
||||
WebRtc_UWord8& payloadTypeRED,
|
||||
WebRtc_UWord8& payloadTypeFEC) const;
|
||||
WebRtc_Word32 GenericFECStatus(bool& enable,
|
||||
WebRtc_UWord8& payloadTypeRED,
|
||||
WebRtc_UWord8& payloadTypeFEC) const;
|
||||
|
||||
WebRtc_Word32 SetFecParameters(
|
||||
const FecProtectionParams* delta_params,
|
||||
const FecProtectionParams* key_params);
|
||||
WebRtc_Word32 SetFecParameters(
|
||||
const FecProtectionParams* delta_params,
|
||||
const FecProtectionParams* key_params);
|
||||
|
||||
protected:
|
||||
WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType,
|
||||
RtpVideoCodecTypes& videoType);
|
||||
protected:
|
||||
WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType,
|
||||
RtpVideoCodecTypes& videoType);
|
||||
|
||||
private:
|
||||
void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
|
||||
const WebRtc_UWord32 now);
|
||||
private:
|
||||
void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
|
||||
const WebRtc_UWord32 now);
|
||||
|
||||
WebRtc_Word32 SendPaddingAccordingToBitrate(
|
||||
WebRtc_Word8 payload_type,
|
||||
WebRtc_UWord32 capture_timestamp,
|
||||
int64_t capture_time_ms);
|
||||
WebRtc_Word32 SendPaddingAccordingToBitrate(
|
||||
WebRtc_Word8 payload_type,
|
||||
WebRtc_UWord32 capture_timestamp,
|
||||
int64_t capture_time_ms);
|
||||
|
||||
WebRtc_Word32 _id;
|
||||
const bool _audioConfigured;
|
||||
RTPSenderAudio* _audio;
|
||||
RTPSenderVideo* _video;
|
||||
WebRtc_Word32 _id;
|
||||
const bool _audioConfigured;
|
||||
RTPSenderAudio* _audio;
|
||||
RTPSenderVideo* _video;
|
||||
|
||||
CriticalSectionWrapper* _sendCritsect;
|
||||
CriticalSectionWrapper* _sendCritsect;
|
||||
|
||||
CriticalSectionWrapper* _transportCritsect;
|
||||
Transport* _transport;
|
||||
Transport* _transport;
|
||||
bool _sendingMedia;
|
||||
|
||||
bool _sendingMedia;
|
||||
WebRtc_UWord16 _maxPayloadLength;
|
||||
WebRtc_UWord16 _targetSendBitrate;
|
||||
WebRtc_UWord16 _packetOverHead;
|
||||
|
||||
WebRtc_UWord16 _maxPayloadLength;
|
||||
WebRtc_UWord16 _targetSendBitrate;
|
||||
WebRtc_UWord16 _packetOverHead;
|
||||
WebRtc_Word8 _payloadType;
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
|
||||
|
||||
WebRtc_Word8 _payloadType;
|
||||
std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
|
||||
RtpHeaderExtensionMap _rtpHeaderExtensionMap;
|
||||
WebRtc_Word32 _transmissionTimeOffset;
|
||||
|
||||
RtpHeaderExtensionMap _rtpHeaderExtensionMap;
|
||||
WebRtc_Word32 _transmissionTimeOffset;
|
||||
// NACK
|
||||
WebRtc_UWord32 _nackByteCountTimes[NACK_BYTECOUNT_SIZE];
|
||||
WebRtc_Word32 _nackByteCount[NACK_BYTECOUNT_SIZE];
|
||||
Bitrate _nackBitrate;
|
||||
|
||||
// NACK
|
||||
WebRtc_UWord32 _nackByteCountTimes[NACK_BYTECOUNT_SIZE];
|
||||
WebRtc_Word32 _nackByteCount[NACK_BYTECOUNT_SIZE];
|
||||
Bitrate _nackBitrate;
|
||||
RTPPacketHistory* _packetHistory;
|
||||
TransmissionBucket _sendBucket;
|
||||
WebRtc_Word64 _timeLastSendToNetworkUpdate;
|
||||
bool _transmissionSmoothing;
|
||||
|
||||
RTPPacketHistory* _packetHistory;
|
||||
TransmissionBucket _sendBucket;
|
||||
WebRtc_Word64 _timeLastSendToNetworkUpdate;
|
||||
bool _transmissionSmoothing;
|
||||
// Statistics
|
||||
WebRtc_UWord32 _packetsSent;
|
||||
WebRtc_UWord32 _payloadBytesSent;
|
||||
|
||||
// statistics
|
||||
WebRtc_UWord32 _packetsSent;
|
||||
WebRtc_UWord32 _payloadBytesSent;
|
||||
|
||||
// RTP variables
|
||||
bool _startTimeStampForced;
|
||||
WebRtc_UWord32 _startTimeStamp;
|
||||
SSRCDatabase& _ssrcDB;
|
||||
WebRtc_UWord32 _remoteSSRC;
|
||||
bool _sequenceNumberForced;
|
||||
WebRtc_UWord16 _sequenceNumber;
|
||||
WebRtc_UWord16 _sequenceNumberRTX;
|
||||
bool _ssrcForced;
|
||||
WebRtc_UWord32 _ssrc;
|
||||
WebRtc_UWord32 _timeStamp;
|
||||
WebRtc_UWord8 _CSRCs;
|
||||
WebRtc_UWord32 _CSRC[kRtpCsrcSize];
|
||||
bool _includeCSRCs;
|
||||
bool _RTX;
|
||||
WebRtc_UWord32 _ssrcRTX;
|
||||
// RTP variables
|
||||
bool _startTimeStampForced;
|
||||
WebRtc_UWord32 _startTimeStamp;
|
||||
SSRCDatabase& _ssrcDB;
|
||||
WebRtc_UWord32 _remoteSSRC;
|
||||
bool _sequenceNumberForced;
|
||||
WebRtc_UWord16 _sequenceNumber;
|
||||
WebRtc_UWord16 _sequenceNumberRTX;
|
||||
bool _ssrcForced;
|
||||
WebRtc_UWord32 _ssrc;
|
||||
WebRtc_UWord32 _timeStamp;
|
||||
WebRtc_UWord8 _CSRCs;
|
||||
WebRtc_UWord32 _CSRC[kRtpCsrcSize];
|
||||
bool _includeCSRCs;
|
||||
bool _RTX;
|
||||
WebRtc_UWord32 _ssrcRTX;
|
||||
};
|
||||
} // namespace webrtc
|
||||
|
||||
|
||||
@ -78,12 +78,11 @@ class RtpSenderTest : public ::testing::Test {
|
||||
protected:
|
||||
RtpSenderTest()
|
||||
: fake_clock_(),
|
||||
rtp_sender_(new RTPSender(0, false, &fake_clock_)),
|
||||
transport_(),
|
||||
rtp_sender_(new RTPSender(0, false, &fake_clock_, &transport_, NULL)),
|
||||
kMarkerBit(true),
|
||||
kType(kRtpExtensionTransmissionTimeOffset),
|
||||
packet_() {
|
||||
EXPECT_EQ(0, rtp_sender_->SetSequenceNumber(kSeqNum));
|
||||
rtp_sender_->SetSequenceNumber(kSeqNum);
|
||||
}
|
||||
~RtpSenderTest() {
|
||||
delete rtp_sender_;
|
||||
@ -199,8 +198,6 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
|
||||
}
|
||||
|
||||
TEST_F(RtpSenderTest, NoTrafficSmoothing) {
|
||||
EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
|
||||
|
||||
WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,
|
||||
kPayload,
|
||||
kMarkerBit,
|
||||
@ -218,9 +215,8 @@ TEST_F(RtpSenderTest, NoTrafficSmoothing) {
|
||||
|
||||
TEST_F(RtpSenderTest, TrafficSmoothing) {
|
||||
rtp_sender_->SetTransmissionSmoothingStatus(true);
|
||||
EXPECT_EQ(0, rtp_sender_->SetStorePacketsStatus(true, 10));
|
||||
rtp_sender_->SetStorePacketsStatus(true, 10);
|
||||
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
|
||||
EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
|
||||
rtp_sender_->SetTargetSendBitrate(300000);
|
||||
|
||||
WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user