| /external/skia/src/image/ |
| SkImage_Gpu.cpp | 86 return GrPixelConfigIsOpaque(fTexture->config()) || fAlphaType == kOpaque_SkAlphaType;
112 GrPixelConfig config = SkImageInfo2GrPixelConfig(info.colorType(), info.alphaType(), local
119 if (!fTexture->readPixels(srcX, srcY, info.width(), info.height(), config,
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| /external/skia/src/ports/ |
| SkFontMgr_fontconfig.cpp | 209 // Note that this config is only used for FcFontRenderPrepare, which we don't even want.
211 SkAutoFcConfig config; local
213 SkAutoFcPattern match(FcFontSetMatch(config, fontSets, SK_ARRAY_COUNT(fontSets),
601 /** Takes control of the reference to 'config'. */
602 explicit SkFontMgr_fontconfig(FcConfig* config)
603 : fFC(config ? config : FcInitLoadConfigAndFonts())
607 // Hold the lock while unrefing the config.
[all...] |
| /external/tinyalsa/ |
| tinycap.c | 188 struct pcm_config config; local
194 memset(&config, 0, sizeof(config));
195 config.channels = channels;
196 config.rate = rate;
197 config.period_size = period_size;
198 config.period_count = period_count;
199 config.format = format;
200 config.start_threshold = 0;
201 config.stop_threshold = 0
[all...] |
| /external/tinycompress/ |
| cplay.c | 202 struct compr_config config; local
242 config.fragment_size = buffer_size/frag;
243 config.fragments = frag;
246 config.fragment_size = 0;
247 config.fragments = 0;
249 config.codec = &codec;
251 compress = compress_open(card, device, COMPRESS_IN, &config);
260 size = config.fragment_size;
261 buffer = malloc(size * config.fragments);
268 num_read = fread(buffer, 1, size * config.fragments, file)
[all...] |
| /external/v8/src/compiler/ |
| register-allocator-verifier.h | 19 RegisterAllocatorVerifier(Zone* zone, const RegisterConfiguration* config,
61 const RegisterConfiguration* config() { return config_; } function in class:v8::internal::compiler::final
|
| /external/webrtc/talk/app/webrtc/ |
| datachannel_unittest.cc | 376 webrtc::InternalDataChannelInit config; local
377 config.id = 1;
378 config.negotiated = true;
379 config.open_handshake_role = webrtc::InternalDataChannelInit::kNone;
383 &provider_, cricket::DCT_SCTP, "test1", config);
392 webrtc::InternalDataChannelInit config; local
393 config.id = 1;
394 config.negotiated = true;
395 config.open_handshake_role = webrtc::InternalDataChannelInit::kAcker;
399 &provider_, cricket::DCT_SCTP, "test1", config);
[all...] |
| /external/webrtc/talk/app/webrtc/test/ |
| peerconnectiontestwrapper.cc | 91 webrtc::PeerConnectionInterface::RTCConfiguration config; local
94 config.servers.push_back(ice_server);
99 config, constraints, std::move(port_allocator),
|
| /external/webrtc/webrtc/audio/ |
| audio_receive_stream.cc | 36 bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) {
37 if (!config.rtp.transport_cc) {
40 for (const auto& extension : config.rtp.extensions) {
49 std::string AudioReceiveStream::Config::Rtp::ToString() const {
65 std::string AudioReceiveStream::Config::ToString() const {
85 const webrtc::AudioReceiveStream::Config& config,
87 : config_(config),
98 channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc)
243 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { function in class:webrtc::internal::AudioReceiveStream
[all...] |
| audio_send_stream.cc | 31 std::string AudioSendStream::Config::Rtp::ToString() const {
47 std::string AudioSendStream::Config::ToString() const {
51 // TODO(solenberg): Encoder config.
60 const webrtc::AudioSendStream::Config& config,
63 : config_(config), audio_state_(audio_state) {
76 channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
77 channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
79 for (const auto& extension : config.rtp.extensions) {
208 const webrtc::AudioSendStream::Config& AudioSendStream::config() const function in class:webrtc::internal::AudioSendStream
[all...] |
| audio_send_stream_unittest.cc | 67 AudioState::Config config; local
68 config.voice_engine = &voice_engine_;
69 audio_state_ = AudioState::Create(config);
107 AudioSendStream::Config& config() { return stream_config_; } function in struct:webrtc::test::__anon27562::ConfigHelper
159 AudioSendStream::Config stream_config_;
169 AudioSendStream::Config config(nullptr);
170 config.rtp.ssrc = kSsrc
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| rent_a_codec.cc | 183 AudioEncoderCopyRed::Config config; local
184 config.payload_type = red_payload_type;
185 config.speech_encoder = encoder;
186 return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCopyRed(config));
195 AudioEncoderCng::Config config; local
196 config.num_channels = encoder->NumChannels();
197 config.payload_type = payload_type;
198 config.speech_encoder = encoder
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
| audio_encoder_copy_red_unittest.cc | 40 AudioEncoderCopyRed::Config config; local
41 config.payload_type = red_payload_type_;
42 config.speech_encoder = &mock_encoder_;
43 red_.reset(new AudioEncoderCopyRed(config));
326 AudioEncoderCopyRed::Config config; local
327 config.speech_encoder = NULL;
328 EXPECT_DEATH(red = new AudioEncoderCopyRed(config),
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_performance_test.cc | 41 NetEq::Config config; local
42 config.sample_rate_hz = kSampRateHz;
43 NetEq* neteq = NetEq::Create(config);
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| TwoWayCommunication.cc | 37 AudioCodingModule::Config config; local
39 config.neteq_config.playout_mode = kPlayoutFax;
40 config.id = 2;
41 _acmB.reset(AudioCodingModule::Create(config));
42 config.id = 4;
43 _acmRefB.reset(AudioCodingModule::Create(config));
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| echo_control_mobile_impl.cc | 390 AecmConfig config; local
391 config.cngMode = comfort_noise_enabled_;
392 config.echoMode = MapSetting(routing_mode_);
394 return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
|
| /external/webrtc/webrtc/modules/audio_processing/test/android/apmtest/jni/ |
| main.c | 80 EGLConfig config; local
91 eglChooseConfig(display, attribs, &config, 1, &numConfigs);
97 eglGetConfigAttrib(display, config, EGL_NATIVE_VISUAL_ID, &format);
101 surface = eglCreateWindowSurface(display, config, engine->app->window, NULL);
102 context = eglCreateContext(display, config, NULL, NULL);
|
| /external/webrtc/webrtc/modules/video_coding/codecs/tools/ |
| video_quality_measurement.cc | 146 int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
152 config->name = FLAGS_test_name;
153 config->description = FLAGS_test_description;
164 config->input_filename = FLAGS_input_filename;
174 config->output_dir = FLAGS_output_dir;
192 config->output_filename = FLAGS_output_filename;
194 config->output_filename = FLAGS_output_dir + "/" + FLAGS_output_filename;
196 test_file = fopen(config->output_filename.c_str(), "wb");
199 config->output_filename.c_str());
205 config->use_single_core = FLAGS_use_single_core
478 webrtc::test::TestConfig config; local
[all...] |
| /external/webrtc/webrtc/p2p/client/ |
| httpportallocator.cc | 124 // but for now is done here and added to the initial config. Note any later
133 PortConfiguration* config = new PortConfiguration(hosts, local
136 ConfigReady(config);
201 PortConfiguration* config = new PortConfiguration(hosts, local
218 config->AddRelay(relay_config);
219 ConfigReady(config);
|
| /external/webrtc/webrtc/tools/agc/ |
| agc_harness.cc | 107 webrtc::Config config; local
108 config.Set<ExperimentalAgc>(new ExperimentalAgc(!legacy_agc));
109 AudioProcessing* audioproc = AudioProcessing::Create(config);
|
| /external/webrtc/webrtc/video/ |
| video_capture_input_unittest.cc | 56 webrtc::VideoSendStream::Config(nullptr),
64 Config config; local
|
| /external/wpa_supplicant_8/src/eap_peer/ |
| eap_tls.c | 35 struct eap_peer_config *config = eap_get_config(sm); local
36 if (config == NULL ||
37 ((sm->init_phase2 ? config->private_key2 : config->private_key)
39 (sm->init_phase2 ? config->engine2 : config->engine) == 0)) {
51 if (eap_peer_tls_ssl_init(sm, &data->ssl, config, EAP_TYPE_TLS)) {
54 if (config->engine) {
59 } else if (config->private_key && !config->private_key_passwd
79 struct eap_peer_config *config = eap_get_config(sm); local
106 struct eap_peer_config *config = eap_get_config(sm); local
222 struct eap_peer_config *config = eap_get_config(sm); local
[all...] |
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| aacenc_core.h | 28 #include "config.h"
51 AACENC_CONFIG config; /* Word16 size: 8 */ member in struct:__anon28881
87 void AacInitDefaultConfig(AACENC_CONFIG *config);
98 const AACENC_CONFIG config); /* pre-initialized config struct */
|
| /frameworks/av/services/audiopolicy/common/managerdefinitions/src/ |
| AudioInputDescriptor.cpp | 173 const audio_config_base_t config = { .sample_rate = mSamplingRate, .channel_mask = mChannelMask, local
175 return config;
|
| /frameworks/av/services/audiopolicy/service/ |
| AudioPolicyClientImplLegacy.cpp | 84 audio_config_t config = AUDIO_CONFIG_INITIALIZER; local
85 config.sample_rate = *pSamplingRate;
86 config.format = *pFormat;
87 config.channel_mask = *pChannelMask;
89 config.offload_info = *offloadInfo;
92 status_t status = af->openOutput(module, &output, &config, pDevices,
95 *pSamplingRate = config.sample_rate;
96 *pFormat = config.format;
97 *pChannelMask = config.channel_mask;
99 *((audio_offload_info_t *)offloadInfo) = config.offload_info
199 audio_config_t config = AUDIO_CONFIG_INITIALIZER;; local
[all...] |
| /frameworks/base/core/java/android/security/net/config/ |
| NetworkSecurityTrustManager.java | 17 package android.security.net.config;
49 public NetworkSecurityTrustManager(NetworkSecurityConfig config) {
50 if (config == null) {
51 throw new NullPointerException("config must not be null");
53 mNetworkSecurityConfig = config;
55 TrustedCertificateStoreAdapter certStore = new TrustedCertificateStoreAdapter(config);
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