/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 178 int NetEqImpl::RegisterPayloadType(enum NetEqDecoder codec, argument 181 LOG_API2(static_cast<int>(rtp_payload_type), codec); local 182 int ret = decoder_database_->RegisterPayload(rtp_payload_type, codec); 184 LOG_FERR2(LS_WARNING, RegisterPayload, rtp_payload_type, codec); 204 enum NetEqDecoder codec, 207 LOG_API2(static_cast<int>(rtp_payload_type), codec); local 213 const int sample_rate_hz = AudioDecoder::CodecSampleRateHz(codec); 214 int ret = decoder_database_->InsertExternal(rtp_payload_type, codec, 217 LOG_FERR2(LS_WARNING, InsertExternal, rtp_payload_type, codec); 370 // Set to wait for new codec 203 RegisterExternalDecoder(AudioDecoder* decoder, enum NetEqDecoder codec, uint8_t rtp_payload_type) argument [all...] |
/external/chromium_org/chrome/renderer/media/ |
H A D | cast_rtp_stream.cc | 184 config->codec = media::cast::CODEC_AUDIO_OPUS; 226 config->codec = media::cast::CODEC_VIDEO_VP8; 229 config->codec = media::cast::CODEC_VIDEO_H264;
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/external/chromium_org/content/renderer/media/android/ |
H A D | media_source_delegate.cc | 435 audio_stream_->audio_decoder_config().codec()) { 730 configs->audio_codec = config.codec(); 741 configs->video_codec = config.codec();
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/external/chromium_org/media/cast/test/ |
H A D | fake_media_source.cc | 110 AVCodecContext* av_codec_context = av_stream->codec; 114 LOG(ERROR) << "Cannot find decoder for the codec: " 125 LOG(ERROR) << "Cannot open AVCodecContext for the codec: " 586 return av_audio_stream()->codec; 590 return av_video_stream()->codec;
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H A D | end2end_unittest.cc | 324 // We need to "decode" the encoded audio frame. The codec is simply to 484 audio_sender_config_.codec = audio_codec; 495 audio_receiver_config_.codec = audio_sender_config_.codec; 516 video_sender_config_.codec = video_codec; 527 video_receiver_config_.codec = video_sender_config_.codec; 862 // Audio test without packet loss using raw PCM 16 audio "codec"; 880 // This tests our Opus audio codec without video.
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H A D | sender.cc | 72 audio_config.codec = media::cast::CODEC_AUDIO_OPUS; 98 video_config.codec = media::cast::CODEC_VIDEO_VP8;
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/external/chromium_org/media/formats/mp4/ |
H A D | mp4_stream_parser.cc | 193 // TODO(strobe): When codec reconfigurations are supported, detect and send 194 // a codec reconfiguration for fragments using a sample description index 234 AudioCodec codec = kUnknownAudioCodec; local 241 codec = kCodecAAC; 268 codec, sample_format, channel_layout, sample_per_second,
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/external/chromium_org/third_party/libjingle/source/talk/media/sctp/ |
H A D | sctpdataengine.cc | 35 #include "talk/media/base/codec.h" 274 cricket::DataCodec codec(kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, 0); 275 codec.SetParam(kCodecParamPort, kSctpDefaultPort); 276 codecs_.push_back(codec); 859 // Puts the specified |param| from the codec identified by |id| into |dest|
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | neteq_rtpplay.cc | 94 "codec"); 99 std::string CodecName(webrtc::NetEqDecoder codec); 132 // We have already printed the codec map. Just end the program. 305 // Maps a codec type to a printable name string. 306 std::string CodecName(webrtc::NetEqDecoder codec) { argument 307 switch (codec) {
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/external/chromium_org/chrome/browser/bookmarks/ |
H A D | bookmark_html_writer.cc | 235 // Converts a time string written to the JSON codec into a time_t string 438 BookmarkCodec codec; local 442 new Writer(codec.Encode(BookmarkModelFactory::GetForProfile(
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcvoiceengine.h | 139 bool FindCodec(const AudioCodec& codec); 140 bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
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/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
H A D | vie_autotest_network.cc | 124 EXPECT_EQ(0, ViE.codec->GetSendCodec( 127 EXPECT_EQ(0, ViE.codec->SetSendCodec(
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H A D | vie_autotest_base.cc | 69 ::TestI420CallSetup(interfaces.codec, interfaces.video_engine,
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/external/libvorbis/ |
H A D | libvorbis.spec | 77 %{_includedir}/vorbis/codec.h
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/external/bluetooth/bluedroid/bta/hf_client/ |
H A D | bta_hf_client_int.h | 162 tBTM_SCO_CODEC_TYPE negotiated_codec; /* negotiated codec */ 260 extern void bta_hf_client_send_at_bcs(UINT32 codec);
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/external/bluetooth/bluedroid/bta/include/ |
H A D | bta_ag_api.h | 116 /* HFP peer supported codec masks */ 233 #define BTA_AG_WBS_EVT 31 /* SCO codec info */ 252 #define BTA_AG_AT_BAC_EVT 26 /* avablable codec */ 511 ** Description Specify the codec type to be used for the subsequent 519 BTA_API void BTA_AgSetCodec(UINT16 handle, tBTA_AG_PEER_CODEC codec);
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/external/chromium_org/media/cast/sender/ |
H A D | external_video_encoder_unittest.cc | 118 video_config_.codec = CODEC_VIDEO_VP8;
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H A D | video_encoder_impl_unittest.cc | 80 video_config_.codec = CODEC_VIDEO_VP8;
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/external/chromium_org/media/filters/ |
H A D | ffmpeg_glue_unittest.cc | 238 AVCodecContext* context = glue_->format_context()->streams[0]->codec;
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/external/chromium_org/remoting/client/plugin/ |
H A D | media_source_video_renderer.cc | 154 switch (config.video_config().codec) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | lattice_neon.S | 14 @ filter routine for iSAC codec, optimized for ARM Neon platform.
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H A D | lpc_masking_model_neon.S | 12 @ iSAC codec, optimized for ARM Neon platform. Reference code in
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | test_callbacks.cc | 73 rtpInfo.type.Video.codec = _codecType;
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | transmit_mixer.cc | 311 CodecInst codec; local 312 channel->GetSendCodec(codec); 313 *max_sample_rate = std::max(*max_sample_rate, codec.plfreq); 314 *max_channels = std::max(*max_channels, codec.channels); 356 // Only bother swapping if we're using a stereo codec.
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/external/chromium_org/ui/gfx/ |
H A D | gfx.target.darwin-arm64.mk | 46 ui/gfx/codec/jpeg_codec.cc \ 47 ui/gfx/codec/png_codec.cc \
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