/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
receive_statistics_unittest.cc | 45 receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); 47 receive_statistics_->IncomingPacket(header2_, kPacketSize2, false); 50 receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); 52 receive_statistics_->IncomingPacket(header2_, kPacketSize2, false); 77 receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); 79 receive_statistics_->IncomingPacket(header2_, kPacketSize2, false); 100 receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); 103 receive_statistics_->IncomingPacket(header2_, kPacketSize2, false); 120 receive_statistics_->IncomingPacket(header1_, kPacketSize1, false); 136 receive_statistics_->IncomingPacket(header1_, kPacketSize1, false) [all...] |
receive_statistics_impl.h | 45 void IncomingPacket(const RTPHeader& rtp_header, 105 void IncomingPacket(const RTPHeader& header,
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
remote_bitrate_estimator_abs_send_time_unittest.cc | 97 IncomingPacket(0, 1000, clock_.TimeInMilliseconds(), 0, 0, true); 113 IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000), 121 IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000), 139 IncomingPacket(0, 1000, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000), 143 IncomingPacket(0, 100, now_ms, 90 * now_ms, AbsSendTime(now_ms, 1000), 164 IncomingPacket(0, 1000, now_ms, 90 * send_time_ms, 174 IncomingPacket(0, 1000, now_ms, send_time_ms, 194 IncomingPacket(0, 1000, now_ms, 90 * send_time_ms, 213 IncomingPacket(0, 1000, now_ms, 90 * send_time_ms, 231 IncomingPacket(0, 1000, now_ms, 90 * send_time_ms [all...] |
remote_estimator_proxy_unittest.cc | 35 void IncomingPacket(uint16_t seq, int64_t time_ms) { 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(kBaseSeq + 2, kBaseTimeMs + 2) [all...] |
remote_bitrate_estimator_unittest_helper.cc | 221 void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc, 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(kDefaultSsrc, 1000, clock_.TimeInMilliseconds() [all...] |
remote_bitrate_estimator_single_stream.h | 30 void IncomingPacket(int64_t arrival_time_ms,
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remote_estimator_proxy.h | 40 void IncomingPacket(int64_t arrival_time_ms,
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remote_bitrate_estimator_unittest_helper.h | 166 // estimator (all other fields are cleared) and call IncomingPacket on the 168 void IncomingPacket(uint32_t ssrc,
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/mock/ |
mock_remote_bitrate_estimator.h | 25 MOCK_METHOD4(IncomingPacket, void(int64_t, size_t, const RTPHeader&, bool));
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/external/webrtc/webrtc/modules/video_coding/test/ |
receiver_tests.h | 34 return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
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/external/webrtc/webrtc/modules/audio_coding/neteq/mock/ |
mock_external_decoder_pcm16b.h | 62 ON_CALL(*this, IncomingPacket(_, _, _, _, _)) 63 .WillByDefault(Invoke(&real_, &ExternalPcm16B::IncomingPacket)); 81 MOCK_METHOD5(IncomingPacket,
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mock_audio_decoder.h | 30 MOCK_METHOD5(IncomingPacket, int(const uint8_t*, size_t, uint16_t, uint32_t,
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
receive_statistics.h | 56 virtual void IncomingPacket(const RTPHeader& rtp_header, 85 void IncomingPacket(const RTPHeader& rtp_header,
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
audio_decoder_isac_t.h | 31 int IncomingPacket(const uint8_t* payload,
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audio_decoder_isac_t_impl.h | 76 int AudioDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
audio_decoder_impl.h | 39 int IncomingPacket(const uint8_t* payload,
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/external/webrtc/webrtc/modules/audio_coding/test/ |
PacketLossTest.h | 26 bool IncomingPacket() override;
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PacketLossTest.cc | 41 bool ReceiverWithPacketLoss::IncomingPacket() { 57 _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
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EncodeDecodeTest.h | 77 virtual bool IncomingPacket();
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EncodeDecodeTest.cc | 182 bool Receiver::IncomingPacket() { 198 EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, 231 EXPECT_TRUE(IncomingPacket());
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/ |
remote_bitrate_estimator.h | 77 virtual void IncomingPacket(int64_t arrival_time_ms,
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
audio_decoder.h | 72 virtual int IncomingPacket(const uint8_t* payload,
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audio_decoder.cc | 60 int AudioDecoder::IncomingPacket(const uint8_t* payload,
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/estimators/ |
remb.cc | 86 recv_stats_->IncomingPacket(media_packet.header(), 97 estimator_->IncomingPacket(arrival_time_ms, media_packet.payload_size(),
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/external/webrtc/webrtc/modules/audio_coding/include/ |
audio_coding_module.h | 516 // int32_t IncomingPacket() 529 virtual int32_t IncomingPacket(const uint8_t* incoming_payload,
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