| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| filemediaengine.h | 144 virtual bool RegisterVoiceProcessor(uint32 ssrc,
149 virtual bool UnregisterVoiceProcessor(uint32 ssrc,
207 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
210 virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
220 virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
223 virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
227 virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
230 virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) {
242 virtual bool RemoveSendStream(uint32 ssrc);
244 virtual bool RemoveRecvStream(uint32 ssrc) { return true;
[all...] |
| voiceprocessor.h | 50 virtual void OnFrame(uint32 ssrc,
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| rtpdataengine.cc | 180 << "' with ssrc=" << stream.first_ssrc()
186 // TODO(pthatcher): This should be per-stream, not per-ssrc.
193 << "' with ssrc=" << stream.first_ssrc();
197 bool RtpDataMediaChannel::RemoveSendStream(uint32 ssrc) {
199 if (!GetStreamBySsrc(send_streams_, ssrc, &found_stream)) {
203 RemoveStreamBySsrc(&send_streams_, ssrc);
204 delete rtp_clock_by_send_ssrc_[ssrc];
205 rtp_clock_by_send_ssrc_.erase(ssrc);
217 << "' with ssrc=" << stream.first_ssrc()
224 << "' with ssrc=" << stream.first_ssrc()
[all...] |
| hybridvideoengine.cc | 108 bool HybridVideoMediaChannel::SetRenderer(uint32 ssrc,
112 ret = channel1_->SetRenderer(ssrc, renderer);
115 ret = channel2_->SetRenderer(ssrc, renderer);
131 bool HybridVideoMediaChannel::MuteStream(uint32 ssrc, bool muted) {
134 ret = channel1_->MuteStream(ssrc, muted);
137 ret = channel2_->MuteStream(ssrc, muted);
175 bool HybridVideoMediaChannel::SetSendStreamFormat(uint32 ssrc,
177 return active_channel_ && active_channel_->SetSendStreamFormat(ssrc, format);
222 bool HybridVideoMediaChannel::SetCapturer(uint32 ssrc,
226 ret = channel1_->SetCapturer(ssrc, capturer)
[all...] |
| filemediaengine.cc | 134 void SetSendSsrc(uint32 ssrc);
141 // Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_.
219 void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) {
221 rtp_dump_reader_->SetSsrc(ssrc);
254 uint32 ssrc; local
255 if (!packet->GetRtpSsrc(&ssrc)) {
260 first_ssrc_ = ssrc;
262 if (ssrc == first_ssrc_) {
310 bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) {
311 if (ssrc != send_ssrc_
[all...] |
| rtputils_unittest.cc | 60 // PT = 206, FMT = 1, Sender SSRC = 0x1111, Media SSRC = 0x1111
67 // PT = 204, SSRC = 0x1111
97 uint32 ssrc; local
98 EXPECT_TRUE(GetRtpSsrc(kPcmuFrame, sizeof(kPcmuFrame), &ssrc));
99 EXPECT_EQ(1u, ssrc);
106 EXPECT_EQ(1u, header.ssrc);
111 EXPECT_FALSE(GetRtpSsrc(kInvalidPacket, sizeof(kInvalidPacket), &ssrc));
135 EXPECT_EQ(3333u, header.ssrc);
151 EXPECT_EQ(3333u, header.ssrc);
182 uint32 ssrc; local
[all...] |
| /external/chromium_org/media/cast/transport/pacing/ |
| mock_paced_packet_sender.h | 23 MOCK_METHOD2(SendRtcpPacket, bool(unsigned int ssrc, PacketRef packet));
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| /external/chromium_org/third_party/webrtc/modules/pacing/include/mock/ |
| mock_paced_sender.h | 26 uint32_t ssrc,
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| receive_statistics.h | 59 virtual void FecPacketReceived(uint32_t ssrc) = 0;
65 // Returns a pointer to the statistician of an ssrc.
66 virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
85 virtual void FecPacketReceived(uint32_t ssrc) OVERRIDE;
87 virtual StreamStatistician* GetStatistician(uint32_t ssrc) const OVERRIDE;
|
| remote_ntp_time_estimator.h | 34 // |ssrc|. The RTCP SR is read from |rtp_rtcp|.
35 bool UpdateRtcpTimestamp(uint32_t ssrc, RtpRtcp* rtp_rtcp);
|
| /external/chromium_org/third_party/webrtc/ |
| video_receive_stream.h | 65 ssrc(0) {}
72 uint32_t ssrc; member in struct:webrtc::VideoReceiveStream::Stats
97 // Sender SSRC used for sending RTCP (such as receiver reports).
124 Rtx() : ssrc(0), payload_type(0) {}
127 uint32_t ssrc; member in struct:webrtc::VideoReceiveStream::Config::Rtp::Rtx
|
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/test/ |
| rtp_header_parser.cc | 30 ssrc(0),
66 uint32 rtp_timestamp, ssrc; local
68 big_endian_reader.ReadU32(&ssrc);
76 parsed_packet->ssrc = ssrc;
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| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| statscollector.h | 67 void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
71 void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
83 // Prepare an SSRC report for the given ssrc. Used internally
85 StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport,
87 // Prepare an SSRC report for the given remote ssrc. Used internally.
88 StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport,
90 // Extracts the ID of a Transport belonging to an SSRC. Used internally.
123 // Helper method to get the id for the track identified by ssrc
[all...] |
| mediastreamsignaling.h | 69 uint32 ssrc) = 0;
74 uint32 ssrc) = 0;
87 uint32 ssrc) = 0;
92 uint32 ssrc) = 0;
97 uint32 ssrc) = 0;
127 // session description. This will set the SSRC used for sending data on
130 // session description. If the DataChannel label and a SSRC is included in
131 // the description, the DataChannel is updated with SSRC that will be used
133 // 4. When both the local and remote SSRC of a DataChannel is set the state of
138 // session description. If a label and a SSRC of a new DataChannel is foun
288 uint32 ssrc; member in struct:webrtc::MediaStreamSignaling::TrackInfo
[all...] |
| /external/chromium_org/third_party/webrtc/video/ |
| receive_statistics_proxy.cc | 19 ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc,
32 stats_.ssrc = ssrc;
69 uint32_t ssrc) {
77 uint32_t ssrc) {
|
| send_statistics_proxy_unittest.cc | 105 const uint32_t ssrc = *it; local
106 StreamStats& ssrc_stats = expected_.substreams[ssrc];
109 uint32_t offset = ssrc * sizeof(RtcpStatistics);
114 callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc);
154 const uint32_t ssrc = *it; local
156 StreamStats& stats = expected_.substreams[ssrc];
157 uint32_t offset = ssrc * sizeof(StreamStats);
160 observer->FrameCountUpdated(kVideoFrameKey, stats.key_frames, ssrc);
161 observer->FrameCountUpdated(kVideoFrameDelta, stats.delta_frames, ssrc);
173 const uint32_t ssrc = *it local
195 const uint32_t ssrc = *it; local
[all...] |
| receive_statistics_proxy.h | 38 ReceiveStatisticsProxy(uint32_t ssrc,
67 uint32_t ssrc) OVERRIDE;
71 uint32_t ssrc) OVERRIDE;
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| channel.h | 110 bool IsStreamMuted(uint32 ssrc);
121 // Mute sending media on the stream with SSRC |ssrc|
122 // If there is only one sending stream SSRC 0 can be used.
123 bool MuteStream(uint32 ssrc, bool mute);
127 bool RemoveRecvStream(uint32 ssrc);
129 bool RemoveSendStream(uint32 ssrc);
283 virtual bool MuteStream_w(uint32 ssrc, bool mute);
284 bool IsStreamMuted_w(uint32 ssrc);
288 bool RemoveRecvStream_w(uint32 ssrc);
[all...] |
| /external/chromium_org/media/cast/transport/rtp_sender/ |
| rtp_sender.h | 63 uint32 ssrc() const { return config_.ssrc; } function in class:media::cast::transport::RtpSender
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.cc | 82 TEST_F(RtpRtcpAPITest, SSRC) {
84 EXPECT_EQ(test_ssrc, module->SSRC());
118 unsigned int ssrc = 0; local
125 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
127 EXPECT_EQ(1u, ssrc);
133 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
137 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
150 rtx_header.ssrc = kRtxSsrc;
153 rtx_header.ssrc = 0;
155 rtx_header.ssrc = kRtxSsrc
[all...] |
| /external/srtp/include/ |
| rtp.h | 77 struct sockaddr_in addr, unsigned int ssrc);
81 struct sockaddr_in addr, unsigned int ssrc);
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| tmmbr_help.h | 41 uint32_t Ssrc(int i) const {
42 return _data.at(i).ssrc;
65 SetElement() : tmmbr(0), packet_oh(0), ssrc(0) {}
68 uint32_t ssrc; member in class:webrtc::TMMBRSet::SetElement
94 bool IsOwner(const uint32_t ssrc, const uint32_t length) const;
|
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| voe_rtp_rtcp_impl.h | 42 // SSRC
43 virtual int SetLocalSSRC(int channel, unsigned int ssrc);
45 virtual int GetLocalSSRC(int channel, unsigned int& ssrc);
47 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc);
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| rtp_player.cc | 45 RawRtpPacket(const uint8_t* data, uint32_t length, uint32_t ssrc,
50 ssrc_(ssrc),
60 uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtpplayer::RawRtpPacket
98 printf("Throw: %08x:%u\n", packet->ssrc(), packet->seq_num());
108 void SetResendTime(uint32_t ssrc, int16_t resendSeqNum) {
114 if (ssrc == packet->ssrc() && resendSeqNum == packet->seq_num() &&
206 int RegisterSsrc(uint32_t ssrc, LostPackets* lost_packets, Clock* clock) {
207 if (handlers_.count(ssrc) > 0) {
210 DEBUG_LOG1("Registering handler for ssrc=%08x", ssrc)
294 virtual uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtpplayer::SsrcHandlers::Handler
437 uint32_t ssrc = header.ssrc; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
| fakemediastreamsignaling.h | 97 uint32 ssrc) {
101 uint32 ssrc) {
105 uint32 ssrc) {
110 uint32 ssrc) {
126 uint32 ssrc) {
|
