/external/webrtc/talk/app/webrtc/ |
H A D | rtpreceiver.cc | 35 uint32_t ssrc, 39 ssrc_(ssrc), 86 uint32_t ssrc, 88 : id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) { 34 AudioRtpReceiver(AudioTrackInterface* track, uint32_t ssrc, AudioProviderInterface* provider) argument 85 VideoRtpReceiver(VideoTrackInterface* track, uint32_t ssrc, VideoProviderInterface* provider) argument
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H A D | remoteaudiosource.cc | 73 uint32_t ssrc, 77 ret->Initialize(ssrc, provider); 93 void RemoteAudioSource::Initialize(uint32_t ssrc, argument 100 ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this))); 72 Create( uint32_t ssrc, AudioProviderInterface* provider) argument
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | packet_source.h | 37 virtual void SelectSsrc(uint32_t ssrc) { argument 39 ssrc_ = ssrc;
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/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
H A D | psfb.h | 30 void From(uint32_t ssrc) { sender_ssrc_ = ssrc; } argument 31 void To(uint32_t ssrc) { media_ssrc_ = ssrc; } argument
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H A D | rtpfb.h | 30 void From(uint32_t ssrc) { sender_ssrc_ = ssrc; } argument 31 void To(uint32_t ssrc) { media_ssrc_ = ssrc; } argument
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H A D | tmmbn.h | 32 void From(uint32_t ssrc) { argument 33 tmmbn_.SenderSSRC = ssrc; 36 bool WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead);
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H A D | voip_metric.h | 38 void To(uint32_t ssrc) { ssrc_ = ssrc; } argument 43 uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtcp::VoipMetric
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H A D | bye.h | 35 void From(uint32_t ssrc) { sender_ssrc_ = ssrc; } argument
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H A D | dlrr.cc | 49 sub_block.ssrc = ByteReader<uint32_t>::ReadBigEndian(&read_at[0]); 75 ByteWriter<uint32_t>::WriteBigEndian(&write_at[0], sub_block.ssrc); 84 bool Dlrr::WithDlrrItem(uint32_t ssrc, argument 92 block.ssrc = ssrc;
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H A D | dlrr.h | 27 uint32_t ssrc; member in struct:webrtc::rtcp::Dlrr::SubBlock 51 bool WithDlrrItem(uint32_t ssrc, uint32_t last_rr, uint32_t delay_last_rr);
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H A D | receiver_report.h | 35 void From(uint32_t ssrc) { sender_ssrc_ = ssrc; } argument
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H A D | tmmbr.h | 31 void From(uint32_t ssrc) { argument 32 tmmbr_.SenderSSRC = ssrc; 34 void To(uint32_t ssrc) { argument 35 tmmbr_item_.SSRC = ssrc;
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H A D | app.h | 36 void From(uint32_t ssrc) { ssrc_ = ssrc; } argument 42 uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtcp::App
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | ssrc_database.cc | 35 while (true) { // Try until get a new ssrc. 37 uint32_t ssrc = random_.Rand(1u, 0xfffffffe); local 38 if (ssrcs_.insert(ssrc).second) { 39 return ssrc; 44 void SSRCDatabase::RegisterSSRC(uint32_t ssrc) { argument 46 ssrcs_.insert(ssrc); 49 void SSRCDatabase::ReturnSSRC(uint32_t ssrc) { argument 51 ssrcs_.erase(ssrc);
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H A D | forward_error_correction.h | 81 // The ssrc member is needed to ensure we can restore the SSRC field of 91 uint32_t ssrc; // SSRC of the current frame. Must be set for FEC member in class:webrtc::ForwardErrorCorrection::ReceivedPacket
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H A D | rtp_packet_history_unittest.cc | 42 void CreateRtpPacket(uint16_t seq_num, uint32_t ssrc, uint8_t payload, argument 52 array[(*cur_pos)++] = ssrc >> 24; 53 array[(*cur_pos)++] = ssrc >> 16; 54 array[(*cur_pos)++] = ssrc >> 8; 55 array[(*cur_pos)++] = ssrc;
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/external/webrtc/webrtc/video/ |
H A D | vie_remb_unittest.cc | 52 unsigned int ssrc = 1234; local 53 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 77 unsigned int ssrc = 1234; local 78 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 103 unsigned int ssrc[] = { 1234, 5678 }; local 104 std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeo 134 unsigned int ssrc[] = { 1234, 5678 }; local 168 unsigned int ssrc[] = { 1234, 5678 }; local 202 unsigned int ssrc = 1234; local 235 unsigned int ssrc = 1234; local [all...] |
/external/adhd/cras/src/common/ |
H A D | rtp.h | 37 uint32_t ssrc; member in struct:rtp_header 62 uint32_t ssrc; member in struct:rtp_header
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/external/webrtc/talk/media/base/ |
H A D | rtpdump_unittest.cc | 50 uint32_t ssrc; local 61 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc)); 62 EXPECT_EQ(kTestSsrc, ssrc); 132 uint32_t ssrc; local 133 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc)); 134 EXPECT_EQ(kTestSsrc, ssrc); 139 // Rewind the stream and read again with a specified ssrc. 148 uint32_t ssrc; local 149 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc)); 150 EXPECT_EQ(send_ssrc, ssrc); [all...] |
H A D | rtputils.h | 43 uint32_t ssrc; member in struct:cricket::RtpHeader
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H A D | rtputils_unittest.cc | 95 uint32_t ssrc; local 96 EXPECT_TRUE(GetRtpSsrc(kPcmuFrame, sizeof(kPcmuFrame), &ssrc)); 97 EXPECT_EQ(1u, ssrc); 104 EXPECT_EQ(1u, header.ssrc); 109 EXPECT_FALSE(GetRtpSsrc(kInvalidPacket, sizeof(kInvalidPacket), &ssrc)); 129 EXPECT_EQ(3333u, header.ssrc); 160 uint32_t ssrc; local 163 &ssrc)); 166 &ssrc)); 169 &ssrc)); [all...] |
/external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
H A D | bwe_rtp.cc | 60 uint32_t ssrc; local 61 while (ss >> ssrc) { 62 ssrcs.insert(ssrc);
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/external/webrtc/talk/session/media/ |
H A D | currentspeakermonitor.cc | 91 uint32_t ssrc = stream_list_it->first; local 92 active_ssrc_to_level_map[ssrc] = stream_list_it->second; 96 if (ssrc_to_speaking_state_map_.find(ssrc) == 98 ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
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/external/webrtc/webrtc/call/ |
H A D | rtc_event_log2rtp_dump.cc | 46 DEFINE_string(ssrc, 52 // written to the output variable |ssrc|, and true is returned. Otherwise, 57 bool ParseSsrc(std::string str, uint32_t* ssrc) { argument 66 ss >> read_mode >> *ssrc; local
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/external/webrtc/webrtc/modules/pacing/ |
H A D | packet_router.cc | 42 bool PacketRouter::TimeToSendPacket(uint32_t ssrc, argument 48 if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) { 49 return rtp_module->TimeToSendPacket(ssrc, sequence_number,
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