/external/webrtc/webrtc/voice_engine/ |
H A D | network_predictor.cc | 28 void NetworkPredictor::UpdatePacketLossRate(uint8_t loss_rate) { argument 33 static_cast<float>(loss_rate));
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/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | PacketLossTest.cc | 31 int loss_rate, 33 loss_rate_ = loss_rate; 27 Setup(AudioCodingModule *acm, RTPStream *rtpStream, std::string out_file_name, int channels, int loss_rate, int burst_length) argument
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/external/webrtc/webrtc/modules/video_coding/ |
H A D | generic_encoder.h | 32 uint8_t loss_rate; member in struct:webrtc::EncoderParameters
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_network_stats_unittest.cc | 130 void SetPacketLossRate(double loss_rate) { argument 131 packet_loss_interval_ = (loss_rate >= 1e-3 ? 132 static_cast<double>(kFrameSizeMs) / loss_rate : 0xffffffff);
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | neteq_quality_test.h | 42 UniformLoss(double loss_rate); 44 void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; } argument
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H A D | neteq_quality_test.cc | 168 // to achieve the target packet loss rate |loss_rate|, when a packet is not 171 static double ProbTrans00Solver(int units, double loss_rate, argument 174 return prob_trans_10 / (1.0f - loss_rate) - prob_trans_10; 175 // 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 * 176 // prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10). 188 const double a = (1.0f - loss_rate) / prob_trans_10; 189 const double b = (loss_rate - 1.0f) * (1.0f + 1.0f / prob_trans_10); 264 UniformLoss::UniformLoss(double loss_rate) argument 265 : loss_rate_(loss_rate) { 324 // pi_1 * prob_trans_10_ ^ (units - 1) == 1 - loss_rate 332 double loss_rate = 0.01f * packet_loss_rate_; local [all...] |
/external/libopus/celt/ |
H A D | quant_bands.c | 264 int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate, int lfe) 278 intra_bias = (opus_int32)((budget**delayedIntra*loss_rate)/(C*512)); 261 quant_coarse_energy(const CELTMode *m, int start, int end, int effEnd, const opus_val16 *eBands, opus_val16 *oldEBands, opus_uint32 budget, opus_val16 *error, ec_enc *enc, int C, int LM, int nbAvailableBytes, int force_intra, opus_val32 *delayedIntra, int two_pass, int loss_rate, int lfe) argument
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H A D | celt_encoder.c | 74 int loss_rate; member in struct:OpusCustomEncoder 1112 if (st->loss_rate>2) 1114 if (st->loss_rate>4) 1116 if (st->loss_rate>8) 1791 &st->delayedIntra, st->complexity >= 4, st->loss_rate, st->lfe); 2278 st->loss_rate = value;
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
H A D | opus_interface.c | 167 int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) { argument 170 OPUS_SET_PACKET_LOSS_PERC(loss_rate));
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/external/webrtc/webrtc/modules/rtp_rtcp/test/testFec/ |
H A D | test_packet_masks_metrics.cc | 253 double loss_rate = static_cast<double>( local 261 result *= (1.0 - loss_rate); 263 result *= loss_rate; 274 double prob01 = prob10 * (loss_rate / (1.0 - loss_rate)); 280 result = (1.0 - loss_rate); 282 result = loss_rate; 537 float loss_rate = loss_model_[k].average_loss_rate; 541 loss_rate, 846 float loss_rate [all...] |
/external/webrtc/webrtc/modules/video_coding/test/ |
H A D | rtp_player.cc | 324 float loss_rate, 333 loss_rate_(loss_rate), 472 float loss_rate, 488 &packet_source, loss_rate, rtt_ms, reordering)); 320 RtpPlayerImpl(PayloadSinkFactoryInterface* payload_sink_factory, const PayloadTypes& payload_types, Clock* clock, rtc::scoped_ptr<test::RtpFileReader>* packet_source, float loss_rate, int64_t rtt_ms, bool reordering) argument 468 Create(const std::string& input_filename, PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock, const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms, bool reordering) argument
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | audio_coding_module_impl.cc | 495 int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) { argument 498 rent_a_codec_.GetEncoderStack()->SetProjectedPacketLossRate(loss_rate /
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