/external/chromium_org/third_party/webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/ |
VideoDecodeEncodeObserver.java | 14 void incomingRate(int videoChannel, int framerate, int bitrate); 23 void outgoingRate(int videoChannel, int framerate, int bitrate);
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VideoCodecInst.java | 40 public native void setStartBitRate(int bitrate); 42 public native void setMaxBitRate(int bitrate);
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/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
rate_statistics_unittest.cc | 55 uint32_t bitrate = stats_.Rate(now_ms); local 56 EXPECT_EQ(0u, bitrate); 61 if (new_bitrate != bitrate) { 62 // New bitrate must be higher than previous one. 63 EXPECT_GT(new_bitrate, bitrate); 66 EXPECT_NEAR(8000000u, bitrate, 80000u); 69 bitrate = new_bitrate; 74 bitrate = stats_.Rate(now_ms); 75 EXPECT_NEAR(8000000u, bitrate, 80000u); 81 if (new_bitrate != bitrate) { [all...] |
/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/include/mock/ |
mock_remote_bitrate_observer.h | 24 void(const std::vector<unsigned int>& ssrcs, unsigned int bitrate));
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/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/adaptivestreaming/ |
VideoQuality.java | 19 long bitrate; field in class:VideoQuality
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AudioQuality.java | 21 long bitrate; field in class:AudioQuality
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/external/aac/libAACenc/src/ |
bandwidth.h | 99 INT bitrate,
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/external/chromium_org/media/cast/sender/ |
congestion_control_unittest.cc | 74 uint32 bitrate = congestion_control_->GetBitrate( local 78 safe_bitrate / kTargetEmptyBufferFraction, bitrate, safe_bitrate * 0.05); 80 bitrate = congestion_control_->GetBitrate( 84 bitrate, 87 bitrate = congestion_control_->GetBitrate( 91 bitrate, 99 bitrate = congestion_control_->GetBitrate( 103 bitrate, 111 bitrate = congestion_control_->GetBitrate( 115 bitrate, [all...] |
audio_encoder.h | 31 int bitrate,
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congestion_control.h | 32 // Returns the bitrate we should use for the next frame. 43 CongestionControl* NewFixedCongestionControl(uint32 bitrate);
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/external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
bitrate_controller_impl.cc | 30 virtual void OnReceivedEstimatedBitrate(const uint32_t bitrate) OVERRIDE { 31 owner_->OnReceivedEstimatedBitrate(bitrate); 163 // TODO(andresp): This is a ugly way to set start bitrate. 165 // Only change start bitrate if we have exactly one observer. By definition 166 // you can only have one start bitrate, once we have our first estimate we 189 // If not enforcing min bitrate, allow the bandwidth estimation to 220 void BitrateControllerImpl::OnReceivedEstimatedBitrate(const uint32_t bitrate) { 222 bandwidth_estimation_.UpdateReceiverEstimate(bitrate); 258 uint32_t bitrate; local 261 bandwidth_estimation_.CurrentEstimate(&bitrate, &fraction_loss, &rtt) 369 uint32_t bitrate; local [all...] |
bitrate_controller_impl.h | 69 uint32_t bitrate) 71 min_bitrate_(bitrate) { 83 void OnReceivedEstimatedBitrate(const uint32_t bitrate); 92 void OnNetworkChanged(const uint32_t bitrate, 97 void NormalRateAllocation(uint32_t bitrate, 103 void LowRateAllocation(uint32_t bitrate,
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send_side_bandwidth_estimation.h | 10 * FEC and NACK added bitrate is handled outside class 27 void CurrentEstimate(uint32_t* bitrate, uint8_t* loss, uint32_t* rtt) const; 41 void SetSendBitrate(uint32_t bitrate); 50 // min bitrate used during last kBweIncreaseIntervalMs.
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send_side_bandwidth_estimation.cc | 63 void SendSideBandwidthEstimation::SetSendBitrate(uint32_t bitrate) { 64 bitrate_ = bitrate; 66 // Clear last sent bitrate history so the new value can be used directly 81 void SendSideBandwidthEstimation::CurrentEstimate(uint32_t* bitrate, 84 *bitrate = bitrate_; 129 // Only start updating bitrate when receiving receiver blocks. 132 // Loss < 2%: Increase rate by 8% of the min bitrate in the last 134 // Note that by remembering the bitrate over the last second one can 181 // bitrate if it is off by as little as 0.5ms. 189 // bitrate before pushing it [all...] |
/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
bitenc.h | 34 Word32 bitrate; member in struct:BITSTREAMENCODER_INIT
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/external/chromium_org/media/base/ |
data_source.h | 42 // Notify the DataSource of the bitrate of the media. 43 // Values of |bitrate| <= 0 are invalid and should be ignored. 44 virtual void SetBitrate(int bitrate) = 0;
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/external/chromium_org/media/tools/player_x11/ |
data_source_logger.h | 31 virtual void SetBitrate(int bitrate) OVERRIDE;
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data_source_logger.cc | 54 void DataSourceLogger::SetBitrate(int bitrate) { 55 VLOG(1) << "SetBitrate(" << bitrate << ")"; 56 data_source_->SetBitrate(bitrate);
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/external/chromium_org/third_party/webrtc/video_engine/include/ |
vie_codec.h | 37 const unsigned int bitrate) = 0; 62 const unsigned int bitrate) = 0; 147 // Gets the bitrate targeted by the video codec rate control in kbit/s. 149 unsigned int* bitrate) const = 0;
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/external/chromium_org/device/serial/ |
serial_io_handler_posix.cc | 20 bool BitrateToSpeedConstant(int bitrate, speed_t* speed) { 25 switch (bitrate) { 56 // Convert a known nominal speed into an integral bitrate. Returns |true| 58 bool SpeedConstantToBitrate(speed_t speed, int* bitrate) { 61 *bitrate = x; \ 96 int bitrate) { 103 serial.custom_divisor = serial.baud_base / bitrate; 111 speed_t speed = static_cast<speed_t>(bitrate); 249 if (options.bitrate) { 251 if (BitrateToSpeedConstant(options.bitrate, &bitrate_opt)) 382 int bitrate = 0; local [all...] |
/external/chromium_org/media/cast/test/ |
cast_benchmarks.cc | 170 : bitrate(bitrate_), 174 return bitrate >= other.bitrate && latency <= other.latency && 178 return bitrate <= other.bitrate && latency >= other.latency && 184 "%f Mbit/s %f ms %f %% ", bitrate, latency, percent_packet_drop); 187 double bitrate; member in struct:media::cast::MeasuringPoint 234 audio_sender_config_.bitrate = kDefaultAudioEncoderBitrate; 378 scoped_ptr<test::PacketPipe> pipe = test::NewBuffer(65536, p.bitrate); 386 available_bitrate_ = p.bitrate; 565 SearchVariable bitrate; member in struct:media::cast::SearchVector [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
RTPencode.cc | 74 void NetEQTest_GetCodec_and_PT(char * name, webrtc::NetEqDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed); 75 int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels); 78 int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels); 248 int bitrate = 0; local 291 printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]); 399 printf("bitrate : Codec bitrate in bps (only applies to vbr codecs)\n\n"); 422 NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed); 475 bitrate = 32000; 477 "Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n") [all...] |
/external/chromium_org/media/filters/ |
file_data_source.h | 33 virtual void SetBitrate(int bitrate) OVERRIDE;
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/external/chromium_org/third_party/speex/include/speex/ |
speex_header.h | 68 spx_int32_t bitrate; /**< Bit-rate used */ member in struct:SpeexHeader
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
opus_test.h | 34 void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
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