/frameworks/av/media/libstagefright/rtsp/ |
AMPEG2TSAssembler.cpp | 50 return addPacket(source); 53 ARTPAssembler::AssemblyStatus AMPEG2TSAssembler::addPacket(
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AAMRAssembler.h | 51 AssemblyStatus addPacket(const sp<ARTPSource> &source);
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AH263Assembler.h | 49 AssemblyStatus addPacket(const sp<ARTPSource> &source);
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AMPEG2TSAssembler.h | 46 AssemblyStatus addPacket(const sp<ARTPSource> &source);
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AMPEG4AudioAssembler.h | 59 AssemblyStatus addPacket(const sp<ARTPSource> &source);
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AMPEG4ElementaryAssembler.h | 71 AssemblyStatus addPacket(const sp<ARTPSource> &source);
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ARawAudioAssembler.h | 53 AssemblyStatus addPacket(const sp<ARTPSource> &source);
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ARawAudioAssembler.cpp | 50 return addPacket(source); 53 ARTPAssembler::AssemblyStatus ARawAudioAssembler::addPacket(
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AAMRAssembler.cpp | 78 return addPacket(source); 101 ARTPAssembler::AssemblyStatus AAMRAssembler::addPacket(
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AH263Assembler.cpp | 45 AssemblyStatus status = addPacket(source); 52 ARTPAssembler::AssemblyStatus AH263Assembler::addPacket(
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AMPEG4ElementaryAssembler.cpp | 206 ARTPAssembler::AssemblyStatus AMPEG4ElementaryAssembler::addPacket( 410 AssemblyStatus status = addPacket(source);
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AMPEG4AudioAssembler.cpp | 482 AssemblyStatus status = addPacket(source); 489 ARTPAssembler::AssemblyStatus AMPEG4AudioAssembler::addPacket(
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
transport_feedback_adapter.h | 36 void AddPacket(uint16_t sequence_number,
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transport_feedback_adapter.cc | 53 void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
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transport_feedback_adapter_unittest.cc | 108 adapter_->AddPacket(info.sequence_number, info.payload_size,
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
receive_statistics_impl.cc | 66 receive_counters_.transmitted.AddPacket(packet_length, header); 68 receive_counters_.retransmitted.AddPacket(packet_length, header); 175 receive_counters_.fec.AddPacket(packet_length, header);
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rtp_sender.cc | 689 transport_feedback_observer_->AddPacket(options.packet_id, length, true); 949 transport_feedback_observer_->AddPacket(options.packet_id, length, true); [all...] |
/external/webrtc/webrtc/modules/video_coding/test/ |
rtp_player.cc | 97 void AddPacket(RawRtpPacket* packet) { 439 lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
rtp_rtcp_defines.h | 312 virtual void AddPacket(uint16_t sequence_number,
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/external/webrtc/webrtc/ |
common_types.h | 838 void AddPacket(size_t packet_length, const RTPHeader& header) {
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/external/webrtc/webrtc/voice_engine/ |
channel.cc | 65 void AddPacket(uint16_t sequence_number, 71 feedback_observer_->AddPacket(sequence_number, length, was_paced); [all...] |