/external/v8/src/x64/ |
deoptimizer-x64.cc | 119 const RegisterConfiguration* config = RegisterConfiguration::Crankshaft(); local 120 for (int i = 0; i < config->num_allocatable_double_registers(); ++i) { 121 int code = config->GetAllocatableDoubleCode(i); 260 for (int i = 0; i < config->num_allocatable_double_registers(); ++i) { 261 int code = config->GetAllocatableDoubleCode(i);
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/external/valgrind/callgrind/ |
clo.c | 28 #include "config.h" // for VG_PREFIX 86 fn_config* config; member in struct:_config_node 135 node->config = 0; 154 * Returns function config object for pattern <name> 169 if (!node->config) node->config = new_fnc(); 170 return node->config; 210 new_sub->config = new_fnc(); 211 return new_sub->config; 247 new_sub->config = new_fnc() [all...] |
/external/webp/src/enc/ |
alpha_enc.c | 54 WebPConfig config; local 68 WebPConfigInit(&config); 69 config.lossless = 1; 74 config.exact = 1; 75 config.method = effort_level; // impact is very small 79 config.quality = 8.f * effort_level; 80 assert(config.quality >= 0 && config.quality <= 100.f); 86 ok = (VP8LEncodeStream(&config, &picture, bw, 0 /*use_cache*/) == VP8_ENC_OK); 356 const WebPConfig* config = enc->config_ local [all...] |
picture_enc.c | 237 WebPConfig config; local 243 if (!WebPConfigPreset(&config, WEBP_PRESET_DEFAULT, quality_factor) || 248 config.lossless = !!lossless; 256 ok = import(&pic, rgba, stride) && WebPEncode(&config, &pic);
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webp_enc.c | 70 // Mapping from config->method_ to coding tools used. 96 const WebPConfig* const config = enc->config_; local 97 const int method = config->method; 98 const int limit = 100 - config->partition_limit; 112 enc->thread_level_ = config->thread_level; 114 enc->do_search_ = (config->target_size > 0 || config->target_PSNR > 0); 115 if (!config->low_memory) { 144 static VP8Encoder* InitVP8Encoder(const WebPConfig* const config, 148 (config->filter_strength > 0) || (config->autofilter > 0) [all...] |
/external/webrtc/talk/app/webrtc/ |
peerconnectionfactory_unittest.cc | 160 webrtc::PeerConnectionInterface::RTCConfiguration config; local 165 config, nullptr, nullptr, std::move(dtls_identity_store), &observer)); 173 PeerConnectionInterface::RTCConfiguration config; local 176 config.servers.push_back(ice_server); 179 config.servers.push_back(ice_server); 182 config.servers.push_back(ice_server); 186 config, nullptr, std::move(port_allocator_), 206 PeerConnectionInterface::RTCConfiguration config; local 212 config.servers.push_back(ice_server); 216 config, nullptr, std::move(port_allocator_) 234 PeerConnectionInterface::RTCConfiguration config; local 258 PeerConnectionInterface::RTCConfiguration config; local 277 PeerConnectionInterface::RTCConfiguration config; local 310 PeerConnectionInterface::RTCConfiguration config; local [all...] |
/external/webrtc/webrtc/audio/ |
audio_receive_stream_unittest.cc | 78 AudioState::Config config; local 79 config.voice_engine = &voice_engine_; 80 audio_state_ = AudioState::Create(config); 118 AudioReceiveStream::Config& config() { return stream_config_; } function in struct:webrtc::test::__anon36906::ConfigHelper 160 AudioReceiveStream::Config stream_config_; 205 AudioReceiveStream::Config config; local 206 config.rtp.remote_ssrc = kRemoteSsrc [all...] |
/external/webrtc/webrtc/call/ |
bitrate_estimator_tests.cc | 110 AudioState::Config audio_state_config; 112 Call::Config config; local 113 config.audio_state = AudioState::Create(audio_state_config); 114 receiver_call_.reset(Call::Create(config)); 115 sender_call_.reset(Call::Create(config)); 122 video_send_config_ = VideoSendStream::Config(send_transport_.get()); 131 receive_config_ = VideoReceiveStream::Config(receive_transport_.get()); 185 AudioReceiveStream::Config receive_config; 258 VideoReceiveStream::Config receive_config_ [all...] |
rampup_tests.cc | 69 Call::Config RampUpTester::GetSenderCallConfig() { 70 Call::Config call_config; 100 VideoSendStream::Config* send_config, 101 std::vector<VideoReceiveStream::Config>* receive_configs, 154 for (VideoReceiveStream::Config& recv_config : *receive_configs) { 180 AudioSendStream::Config* send_config, 181 std::vector<AudioReceiveStream::Config>* receive_configs) { 202 for (AudioReceiveStream::Config& recv_config : *receive_configs) { 351 Call::Config RampUpDownUpTester::GetReceiverCallConfig() { 352 Call::Config config local [all...] |
/external/webrtc/webrtc/examples/peerconnection/client/ |
conductor.cc | 116 webrtc::PeerConnectionInterface::RTCConfiguration config; local 119 config.servers.push_back(server); 131 config, &constraints, NULL, NULL, this);
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
acm_receiver_unittest_oldapi.cc | 61 AudioCodingModule::Config config; local 62 acm_.reset(new AudioCodingModuleImpl(config)); 63 receiver_.reset(new AcmReceiver(config));
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
audio_encoder_opus.cc | 26 AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) { 27 AudioEncoderOpus::Config config; local 28 config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 48); 29 config.num_channels = codec_inst.channels; 30 config.bitrate_bps = codec_inst.rate; 31 config.payload_type = codec_inst.pltype; 32 config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip 34 return config; [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/ |
audio_decoder_unittest.cc | 293 AudioEncoderPcmU::Config config; local 294 config.frame_size_ms = static_cast<int>(frame_size_ / 8); 295 config.payload_type = payload_type_; 296 audio_encoder_.reset(new AudioEncoderPcmU(config)); 306 AudioEncoderPcmA::Config config; local 307 config.frame_size_ms = static_cast<int>(frame_size_ / 8); 308 config.payload_type = payload_type_; 309 audio_encoder_.reset(new AudioEncoderPcmA(config)); 321 AudioEncoderPcm16B::Config config; local 338 AudioEncoderIlbc::Config config; local 370 AudioEncoderIsac::Config config; local 387 AudioEncoderIsac::Config config; local 404 AudioEncoderIsacFix::Config config; local 423 AudioEncoderG722::Config config; local 440 AudioEncoderG722::Config config; local 455 AudioEncoderOpus::Config config; local 469 AudioEncoderOpus::Config config; local [all...] |
neteq_external_decoder_unittest.cc | 176 NetEq::Config config; local 177 config.sample_rate_hz = 179 neteq_internal_.reset(NetEq::Create(config));
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
neteq_quality_test.cc | 247 NetEq::Config config; local 248 config.sample_rate_hz = out_sampling_khz_ * 1000; 249 neteq_.reset(NetEq::Create(config));
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/external/webrtc/webrtc/modules/audio_processing/ |
echo_cancellation_impl.cc | 445 void EchoCancellationImpl::SetExtraOptions(const Config& config) { 448 extended_filter_enabled_ = config.Get<ExtendedFilter>().enabled; 449 delay_agnostic_enabled_ = config.Get<DelayAgnostic>().enabled; 478 AecConfig config; local 479 config.metricsMode = metrics_enabled_; 480 config.nlpMode = MapSetting(suppression_level_); 481 config.skewMode = drift_compensation_enabled_; 482 config.delay_logging = delay_logging_enabled_; 489 return WebRtcAec_set_config(static_cast<Handle*>(handle), config); [all...] |
gain_control_impl.cc | 423 WebRtcAgcConfig config; local 427 //config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_); 428 config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_); 429 config.compressionGaindB = 431 config.limiterEnable = limiter_enabled_; 433 return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
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/external/webrtc/webrtc/modules/audio_processing/test/ |
debug_dump_test.cc | 32 const StreamConfig& config) { 34 if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() || 35 buffer_ref->num_channels() != config.num_channels()) { 36 buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(), 37 config.num_channels())); 49 const Config& config, 53 explicit DebugDumpGenerator(const Config& config); 85 const StreamConfig& config, 384 Config config; local 472 Config config; local 481 Config config; local 498 Config config; local 510 Config config; local 522 Config config; local 536 Config config; local 551 Config config; local 574 Config config; local 588 Config config; local 602 Config config; local [all...] |
/external/webrtc/webrtc/p2p/base/ |
transportcontroller_unittest.cc | 139 cricket::IceConfig config; local 140 config.receiving_timeout_ms = receiving_timeout_ms; 141 config.gather_continually = gather_continually; 142 return config;
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/external/wpa_supplicant_8/src/eap_peer/ |
eap_tnc.c | 39 struct eap_peer_config *config = eap_get_config(sm); local 45 if (config && config->fragment_size) 46 data->fragment_size = config->fragment_size;
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/external/wpa_supplicant_8/wpa_supplicant/ |
main_winsvc.c | 12 * 'adapter' and 'config' values. 66 TCHAR adapter[TBUFLEN], config[TBUFLEN], ctrl_interface[TBUFLEN]; local 103 buflen = sizeof(config); 104 ret = RegQueryValueEx(hk, TEXT("config"), NULL, NULL, 105 (LPBYTE) config, &buflen); 107 config[sizeof(config) - 1] = '\0'; 108 wpa_unicode2ascii_inplace(config); 109 printf("config[len=%d] '%s'\n", 110 (int) buflen, (char *) config); [all...] |
/frameworks/av/media/libeffects/testlibs/ |
EffectEqualizer.cpp | 111 effect_config_t config; member in struct:android::__anon38144::EqualizerContext 250 pContext->config = *pConfig; 278 *pConfig = pContext->config; 307 pContext->config.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 308 pContext->config.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO; 309 pContext->config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 310 pContext->config.inputCfg.samplingRate = 44100; 311 pContext->config.inputCfg.bufferProvider.getBuffer = NULL; 312 pContext->config.inputCfg.bufferProvider.releaseBuffer = NULL; 313 pContext->config.inputCfg.bufferProvider.cookie = NULL [all...] |
/frameworks/av/media/libstagefright/rtsp/ |
APacketSource.cpp | 225 CHECK(GetAttribute(params, "config", &val)); 227 sp<ABuffer> config = decodeHex(val); local 228 CHECK(config != NULL); 229 CHECK_GE(config->size(), 4u); 231 const uint8_t *data = config->data(); 273 CHECK(GetAttribute(params, "config", &val)); 275 sp<ABuffer> config = decodeHex(val); local 276 CHECK(config != NULL); 280 CHECK_LT(20 + config->size(), 128u); 297 sp<ABuffer> csd = new ABuffer(sizeof(kStaticESDS) + config->size()) 370 sp<ABuffer> config = decodeHex(val); local [all...] |
/frameworks/av/media/libstagefright/wifi-display/ |
VideoFormats.cpp | 281 const config_t *config = &mResolutionTable[type][index]; local 283 if (config->width == 0) { 288 *width = config->width; 292 *height = config->height; 296 *framesPerSecond = config->framesPerSecond; 300 *interlaced = config->interlaced;
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/frameworks/av/radio/ |
IRadio.cpp | 61 virtual status_t setConfiguration(const struct radio_band_config *config) 64 if (config == NULL) { 68 data.write(config, sizeof(struct radio_band_config)); 76 virtual status_t getConfiguration(struct radio_band_config *config) 79 if (config == NULL) { 87 reply.read(config, sizeof(struct radio_band_config)); 238 struct radio_band_config config; local 239 data.read(&config, sizeof(struct radio_band_config)); 240 status_t status = setConfiguration(&config); 246 struct radio_band_config config; local [all...] |