/external/webrtc/webrtc/modules/video_coding/codecs/vp9/ |
H A D | vp9_impl.cc | 70 config_(NULL), 101 if (config_ != NULL) { 102 delete config_; 103 config_ = NULL; 135 config_->ss_target_bitrate[i] = config_->layer_target_bitrate[i] = 136 static_cast<int>(static_cast<int64_t>(config_->rc_target_bitrate) * 157 config_->ss_target_bitrate[i] = static_cast<unsigned int>( 158 config_->rc_target_bitrate * rate_ratio[i] / total); 160 config_ [all...] |
/external/webrtc/webrtc/modules/video_coding/codecs/test/ |
H A D | videoprocessor_unittest.cc | 40 TestConfig config_; member in class:webrtc::test::VideoProcessorTest 48 config_.codec_settings = &codec_settings_; 49 config_.codec_settings->startBitrate = 100; 50 config_.codec_settings->width = 352; 51 config_.codec_settings->height = 288; 71 &packet_manipulator_mock_, config_, &stats_); 83 &packet_manipulator_mock_, config_, &stats_);
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H A D | packet_manipulator.cc | 25 config_(config), 48 config_.packet_size_in_bytes); 59 } else if (RandomUniform() < config_.packet_loss_probability || 63 if (config_.packet_loss_mode == kBurst) { 65 active_burst_packets_ = config_.packet_loss_burst_length - 1;
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H A D | videoprocessor.cc | 53 config_(config), 79 bit_rate_factor_ = config_.codec_settings->maxFramerate * 0.001 * 8; // bits 87 last_encoder_frame_width_ = config_.codec_settings->width; 88 last_encoder_frame_height_ = config_.codec_settings->height; 112 if (!config_.use_single_core) { 116 encoder_->InitEncode(config_.codec_settings, nbr_of_cores, 117 config_.networking_config.max_payload_size_in_bytes); 123 init_result = decoder_->InitDecode(config_.codec_settings, nbr_of_cores); 130 if (config_.verbose) { 136 config_ [all...] |
H A D | videoprocessor_integrationtest.cc | 111 webrtc::test::TestConfig config_; member in class:webrtc::VideoProcessorIntegrationTest 165 config_.input_filename = webrtc::test::ResourcePath("foreman_cif", "yuv"); 168 config_.output_filename = webrtc::test::TempFilename( 170 config_.frame_length_in_bytes = 172 config_.verbose = false; 174 config_.use_single_core = true; 176 config_.keyframe_interval = key_frame_interval_; 177 config_.networking_config.packet_loss_probability = packet_loss_; 180 config_.codec_settings = &codec_settings_; 181 config_ [all...] |
/external/webrtc/webrtc/video/ |
H A D | video_receive_stream.cc | 150 config_(config), 154 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); 157 config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions); 174 vie_channel_->SetProtectionMode(config_.rtp.nack.rtp_history_ms > 0, false, 176 RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff) 179 vie_channel_->SetRTCPMode(config_.rtp.rtcp_mode); 181 RTC_DCHECK(config_.rtp.remote_ssrc != 0); 183 RTC_DCHECK(config_.rtp.local_ssrc != 0); 184 RTC_DCHECK(config_.rtp.remote_ssrc != config_ [all...] |
H A D | send_statistics_proxy_unittest.cc | 26 : fake_clock_(1234), config_(GetTestConfig()), avg_delay_ms_(0), 91 VideoSendStream::Config config_; member in class:webrtc::SendStatisticsProxyTest 101 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); 102 it != config_.rtp.ssrcs.end(); 115 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); 116 it != config_.rtp.rtx.ssrcs.end(); 159 for (std::vector<uint32_t>::const_iterator it = config_.rtp.ssrcs.begin(); 160 it != config_.rtp.ssrcs.end(); 172 for (std::vector<uint32_t>::const_iterator it = config_.rtp.rtx.ssrcs.begin(); 173 it != config_ [all...] |
H A D | video_send_stream.cc | 125 config_(config), 132 LOG(LS_INFO) << "VideoSendStream: " << config_.ToString(); 133 RTC_DCHECK(!config_.rtp.ssrcs.empty()); 173 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { 174 const std::string& extension = config_.rtp.extensions[i].name; 175 int id = config_.rtp.extensions[i].id; 196 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; 197 const bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; 200 config_.rtp.fec.red_payload_type, 201 config_ [all...] |
/external/vboot_reference/utility/ |
H A D | bmpblk_utility.cc | 45 config_.config_filename.clear(); 46 memset(&config_.header, '\0', BMPBLOCK_SIGNATURE_SIZE); 47 config_.images_map.clear(); 48 config_.screens_map.clear(); 49 config_.localizations.clear(); 77 config_.config_filename = filename; 78 config_.images_map.clear(); 79 config_.screens_map.clear(); 80 config_.localizations.clear(); 81 config_ [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
H A D | audio_encoder_opus.cc | 107 static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1); 118 return config_.num_channels; 130 return config_.bitrate_bps; 150 rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels), 157 info.payload_type = config_.payload_type; 164 RTC_CHECK(RecreateEncoderInstance(config_)); 168 auto conf = config_; 174 auto conf = config_; 180 auto conf = config_; 193 auto conf = config_; [all...] |
H A D | audio_encoder_opus.h | 84 ApplicationMode application() const { return config_.application; } 85 bool dtx_enabled() const { return config_.dtx_enabled; } 92 Config config_; member in class:webrtc::final
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/external/autotest/client/deps/glbench/src/ |
H A D | egl_stuff.cc | 20 surface_ = eglCreateWindowSurface(display_, config_, native_window, NULL); 42 if (!config_) { 67 eglChooseConfig(display_, attribs, &config_, 1, &num_configs); 76 eglGetConfigAttrib(display_, config_, EGL_NATIVE_VISUAL_ID, &visual_id); 111 CHECK(config_); 112 return eglCreateContext(display_, config_, NULL, attribs);
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H A D | egl_stuff.h | 16 config_(NULL), 49 EGLConfig config_; member in class:EGLInterface
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H A D | waffle_stuff.h | 16 config_(NULL), 49 struct waffle_config *config_; member in class:WaffleInterface
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/external/webrtc/talk/app/webrtc/ |
H A D | datachannel.cc | 182 config_ = config; 184 switch (config_.open_handshake_role) { 228 return config_.maxRetransmits == -1 && 229 config_.maxRetransmitTime == -1; 296 ASSERT(config_.id < 0 && sid >= 0 && data_channel_type_ == cricket::DCT_SCTP); 297 if (config_.id == sid) { 301 config_.id = sid; 312 if (config_.id >= 0) { 313 provider_->AddSctpDataStream(config_.id); 346 (data_channel_type_ == cricket::DCT_RTP) ? receive_ssrc_ : config_ [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | audio_encoder_ilbc.cc | 45 : config_(config), 127 info.payload_type = config_.payload_type; 134 RTC_CHECK(config_.IsOk()); 136 const int encoder_frame_size_ms = config_.frame_size_ms > 30 137 ? config_.frame_size_ms / 2 138 : config_.frame_size_ms;
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/external/webrtc/webrtc/test/ |
H A D | fake_encoder.cc | 43 config_ = *config; 44 target_bitrate_kbps_ = config_.startBitrate; 51 assert(config_.maxFramerate > 0); 52 int64_t time_since_last_encode_ms = 1000 / config_.maxFramerate; 60 if (time_since_last_encode_ms > 3 * 1000 / config_.maxFramerate) { 63 time_since_last_encode_ms = 3 * 1000 / config_.maxFramerate; 69 config_.simulcastStream[0].minBitrate * time_since_last_encode_ms); 78 assert(config_.numberOfSimulcastStreams > 0); 79 for (unsigned char i = 0; i < config_.numberOfSimulcastStreams; ++i) { 85 config_ [all...] |
H A D | fake_network_pipe.cc | 77 config_(config), 100 config_ = config; // Shallow copy of the struct. 109 if (config_.queue_length_packets > 0 && 110 capacity_link_.size() >= config_.queue_length_packets) { 120 if (config_.link_capacity_kbps > 0) 121 capacity_delay_ms = data_length / (config_.link_capacity_kbps / 8); 166 if (UniformLoss(config_.loss_percent)) { 173 int extra_delay = GaussianRandom(config_.queue_delay_ms, 174 config_.delay_standard_deviation_ms);
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/external/webrtc/talk/media/webrtc/ |
H A D | fakewebrtccall.cc | 40 const webrtc::AudioSendStream::Config& config) : config_(config) { 46 return config_; 73 : config_(config), received_packets_(0) { 79 return config_; 104 config_(config), 112 return config_; 183 if (config_.encoder_settings.payload_name == "VP8") { 186 } else if (config_.encoder_settings.payload_name == "VP9") { 191 << config_.encoder_settings.payload_name; 212 : config_(confi [all...] |
/external/webrtc/webrtc/audio/ |
H A D | audio_state_unittest.cc | 27 config_.voice_engine = &voice_engine_; 29 AudioState::Config& config() { return config_; } 34 AudioState::Config config_; member in struct:webrtc::test::__anon21227::ConfigHelper
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H A D | audio_receive_stream.cc | 87 : config_(config), 90 LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString(); 91 RTC_DCHECK_NE(config_.voe_channel_id, -1); 97 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); 135 LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); 138 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc); 193 stats.remote_ssrc = config_.rtp.remote_ssrc; 198 if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) { 245 return config_;
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H A D | audio_send_stream.cc | 63 : config_(config), audio_state_(audio_state) { 64 LOG(LS_INFO) << "AudioSendStream: " << config_.ToString(); 65 RTC_DCHECK_NE(config_.voe_channel_id, -1); 70 channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id); 94 LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString(); 128 stats.local_ssrc = config_.rtp.ssrc; 146 if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) { 210 return config_;
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/external/opencv3/3rdparty/libwebp/enc/ |
H A D | vp8li.h | 27 const WebPConfig* config_; // user configuration and parameters member in struct:__anon15392
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/external/webp/src/enc/ |
H A D | vp8li.h | 28 const WebPConfig* config_; // user configuration and parameters member in struct:__anon21135
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/external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
H A D | audio_encoder_cng_unittest.cc | 42 config_.speech_encoder = &mock_encoder_; 46 config_.vad = mock_vad_; 47 config_.payload_type = kCngPayloadType; 60 // The config_ parameters may be changed by the TEST_Fs up until CreateCng() 72 cng_.reset(new AudioEncoderCng(config_)); 122 EXPECT_EQ(static_cast<size_t>(config_.num_cng_coefficients + 1), 187 AudioEncoderCng::Config config_; member in class:webrtc::AudioEncoderCngTest 270 if ((i % (config_.sid_frame_interval_ms / 10)) < kBlocksPerFrame) { 274 EXPECT_EQ(static_cast<size_t>(config_.num_cng_coefficients) + 1, 365 EXPECT_EQ(static_cast<size_t>(config_ [all...] |