/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
H A D | remote_estimator_proxy_unittest.cc | 35 void IncomingPacket(uint16_t seq, int64_t time_ms) { function in class:webrtc::RemoteEstimatorProxyTest 40 proxy_.IncomingPacket(time_ms, kDefaultPacketSize, header, true); 62 IncomingPacket(kBaseSeq, kBaseTimeMs); 86 IncomingPacket(kBaseSeq, kBaseTimeMs); 87 IncomingPacket(kBaseSeq + 1, kBaseTimeMs + kMaxSmallDeltaMs); 88 IncomingPacket(kBaseSeq + 2, kBaseTimeMs + (2 * kMaxSmallDeltaMs) + 1); 122 IncomingPacket(kBaseSeq, kBaseTimeMs); 123 IncomingPacket(kBaseSeq + 1, kBaseTimeMs + kTooLargeDelta); 169 IncomingPacket(kBaseSeq, kBaseTimeMs); 170 IncomingPacket(kBaseSe [all...] |
H A D | remote_estimator_proxy.cc | 44 void RemoteEstimatorProxy::IncomingPacket(int64_t arrival_time_ms, function in class:webrtc::RemoteEstimatorProxy
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H A D | remote_bitrate_estimator_abs_send_time.cc | 239 void RemoteBitrateEstimatorAbsSendTime::IncomingPacket(int64_t arrival_time_ms, function in class:webrtc::RemoteBitrateEstimatorAbsSendTime
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H A D | remote_bitrate_estimator_single_stream.cc | 70 void RemoteBitrateEstimatorSingleStream::IncomingPacket(int64_t arrival_time_ms, function in class:webrtc::RemoteBitrateEstimatorSingleStream
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H A D | remote_bitrate_estimator_unittest_helper.cc | 221 void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc, function in class:webrtc::RemoteBitrateEstimatorTest 233 bitrate_estimator_->IncomingPacket(arrival_time + kArrivalTimeClockOffsetMs, 254 // since both are used in IncomingPacket(). 257 IncomingPacket(packet->ssrc, packet->size, 325 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, 334 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, 362 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp, 368 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp, 385 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp, 387 IncomingPacket(kDefaultSsr [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/ |
H A D | audio_decoder.cc | 60 int AudioDecoder::IncomingPacket(const uint8_t* payload, function in class:webrtc::AudioDecoder
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
H A D | audio_decoder_isac_t_impl.h | 76 int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload, function in class:webrtc::AudioDecoderIsacT
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | audio_decoder_impl.cc | 52 int AudioDecoderCng::IncomingPacket(const uint8_t* payload, function in class:webrtc::AudioDecoderCng
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/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | PacketLossTest.cc | 41 bool ReceiverWithPacketLoss::IncomingPacket() { function in class:webrtc::ReceiverWithPacketLoss 57 _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
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H A D | EncodeDecodeTest.cc | 182 bool Receiver::IncomingPacket() { function in class:webrtc::Receiver 198 EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, 231 EXPECT_TRUE(IncomingPacket());
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/external/webrtc/webrtc/test/ |
H A D | fake_network_pipe_unittest.cc | 31 void IncomingPacket(const uint8_t* data, size_t length) { function in class:webrtc::MockReceiver
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | receive_statistics_impl.cc | 52 void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header, function in class:webrtc::StreamStatisticianImpl 392 void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header, function in class:webrtc::ReceiveStatisticsImpl 410 impl->IncomingPacket(header, packet_length, retransmitted); 512 void NullReceiveStatistics::IncomingPacket(const RTPHeader& rtp_header, function in class:webrtc::NullReceiveStatistics
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/external/webrtc/webrtc/modules/video_coding/test/ |
H A D | rtp_player.cc | 252 void IncomingPacket(const uint8_t* data, size_t length) { function in class:webrtc::rtpplayer::SsrcHandlers 445 ssrc_handlers_.IncomingPacket(data, length);
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/external/webrtc/webrtc/modules/video_coding/ |
H A D | video_receiver.cc | 433 int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload, function in class:webrtc::vcm::VideoReceiver
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | audio_coding_module_impl.cc | 653 int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload, function in class:webrtc::acm2::AudioCodingModuleImpl 713 // TODO(kwiberg): Remove this method, and have callers call IncomingPacket 736 IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_);
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