| /external/chromium_org/third_party/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);
160 receive_statistics_->IncomingPacket(header1_, kPacketSize1, false)
[all...] |
| receive_statistics_impl.h | 42 void IncomingPacket(const RTPHeader& rtp_header,
105 virtual void IncomingPacket(const RTPHeader& header,
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| receive_statistics_impl.cc | 69 void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
408 void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header,
426 impl->IncomingPacket(header, bytes, retransmitted);
521 void NullReceiveStatistics::IncomingPacket(const RTPHeader& rtp_header,
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
| mock_external_decoder_pcm16b.h | 66 ON_CALL(*this, IncomingPacket(_, _, _, _, _))
67 .WillByDefault(Invoke(&real_, &ExternalPcm16B::IncomingPacket));
85 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/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| receive_statistics.h | 54 virtual void IncomingPacket(const RTPHeader& rtp_header,
82 virtual void IncomingPacket(const RTPHeader& rtp_header,
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/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 | 76 virtual bool IncomingPacket();
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| EncodeDecodeTest.cc | 183 bool Receiver::IncomingPacket() {
199 EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
232 EXPECT_TRUE(IncomingPacket());
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| receiver_tests.h | 34 return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_unittest_helper.cc | 219 void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
229 bitrate_estimator_->IncomingPacket(arrival_time + kArrivalTimeClockOffsetMs,
250 // since both are used in IncomingPacket().
253 IncomingPacket(packet->ssrc,
323 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
332 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
360 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
366 IncomingPacket(kDefaultSsrc, kMtu, clock_.TimeInMilliseconds(), timestamp,
381 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
383 IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds()
[all...] |
| remote_bitrate_estimator_unittest_helper.h | 165 // estimator (all other fields are cleared) and call IncomingPacket on the
167 void IncomingPacket(uint32_t ssrc,
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| remote_bitrate_estimator_single_stream.cc | 37 virtual void IncomingPacket(int64_t arrival_time_ms,
108 void RemoteBitrateEstimatorSingleStream::IncomingPacket(
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
| BWEStandAlone.cc | 55 _rtpMod->IncomingPacket((uint8_t *) incomingRtpPacket, static_cast<uint16_t>(rtpPacketLength));
64 _rtpMod->IncomingPacket((uint8_t *) incomingRtcpPacket, static_cast<uint16_t>(rtcpPacketLength));
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| audio_decoder_impl.h | 128 virtual int IncomingPacket(const uint8_t* payload,
164 virtual int IncomingPacket(const uint8_t* payload,
265 virtual int IncomingPacket(const uint8_t* payload,
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_isac.h | 82 virtual int IncomingPacket(const uint8_t* payload,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.h | 65 receive_statistics_->IncomingPacket(header, len, false);
|
| /external/chromium_org/third_party/webrtc/test/ |
| fake_network_pipe.h | 66 // Processes the network queues and trigger PacketReceiver::IncomingPacket for
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_receiver.cc | 180 if (vcm_->IncomingPacket(payload_data,
206 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
237 remote_bitrate_estimator_->IncomingPacket(arrival_time_ms,
249 rtp_receive_statistics_->IncomingPacket(
|
| vie_channel_group.cc | 55 virtual void IncomingPacket(int64_t arrival_time_ms,
60 rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
| audio_decoder.h | 97 virtual int IncomingPacket(const uint8_t* payload,
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/include/ |
| remote_bitrate_estimator.h | 75 virtual void IncomingPacket(int64_t arrival_time_ms,
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/tools/ |
| bwe_rtp_play.cc | 85 rbe->IncomingPacket(clock.TimeInMilliseconds(),
|
| /external/chromium_org/third_party/webrtc/video/ |
| rampup_tests.cc | 120 receive_stats_->IncomingPacket(header, length, false);
122 remote_bitrate_estimator_->IncomingPacket(
269 receive_stats_->IncomingPacket(header, length, false);
270 remote_bitrate_estimator_->IncomingPacket(
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