/frameworks/base/services/core/java/com/android/server/storage/ |
FileCollector.java | 42 private static final int AUDIO = 2; 48 AUDIO }) 54 // Audio 55 EXTENSION_MAP.put("aac", AUDIO); 56 EXTENSION_MAP.put("amr", AUDIO); 57 EXTENSION_MAP.put("awb", AUDIO); 58 EXTENSION_MAP.put("snd", AUDIO); 59 EXTENSION_MAP.put("flac", AUDIO); 60 EXTENSION_MAP.put("mp3", AUDIO); 61 EXTENSION_MAP.put("mpga", AUDIO); [all...] |
/external/python/cpython2/Lib/plat-irix5/ |
CL_old.py | 17 # Audio 99 AUDIO = 0 114 UNCOMPRESSED_AUDIO = Algorithm(AUDIO, 0) 115 G711_ULAW = Algorithm(AUDIO, 1) 116 ULAW = Algorithm(AUDIO, 1) 117 G711_ALAW = Algorithm(AUDIO, 2) 118 ALAW = Algorithm(AUDIO, 2) 119 AWARE_MPEG_AUDIO = Algorithm(AUDIO, 3) 120 AWARE_MULTIRATE = Algorithm(AUDIO, 4)
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CD.py | 12 AUDIO = 0
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/external/libmojo/device/bluetooth/ |
bluetooth_common.h | 35 AUDIO,
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/external/python/cpython2/Lib/plat-irix6/ |
CD.py | 12 AUDIO = 0
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/platform_testing/libraries/app-helpers/interfaces/handheld/src/android/platform/helpers/ |
IDownloadsHelper.java | 22 AUDIO, 66 * Setup expectation: Audio is playing 68 * This method will wait for the audio to stop playing or until timeoutInSeconds occur, 71 * @param timeoutInSeconds - timeout value in seconds the test will wait for audio to end
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/toolchain/binutils/binutils-2.27/opcodes/ |
nds32-asm.c | 187 {"a_rt", 15, 5, 0, HW_GPR, NULL}, /* for audio-extension. */ 188 {"a_ru", 10, 5, 0, HW_GPR, NULL}, /* for audio-extension. */ 189 {"a_dx", 9, 1, 0, HW_DXR, NULL}, /* for audio-extension. */ 190 {"a_a30", 16, 4, 0, HW_GPR, parse_a30b20}, /* for audio-extension. */ 191 {"a_b20", 12, 4, 0, HW_GPR, parse_a30b20}, /* for audio-extension. */ 192 {"a_rt21", 5, 7, 0, HW_GPR, parse_rt21}, /* for audio-extension. */ 193 {"a_rte", 5, 7, 0, HW_GPR, parse_rte_start}, /* for audio-extension. */ 194 {"a_rte1", 5, 7, 0, HW_GPR, parse_rte_end}, /* for audio-extension. */ 195 {"a_rte69", 6, 4, 0, HW_GPR, parse_rte69_start}, /* for audio-extension. */ 196 {"a_rte69_1", 6, 4, 0, HW_GPR, parse_rte69_end}, /* for audio-extension. * [all...] |
nds32-asm.h | 138 /* for audio-extension. */ 299 #define AUDIO(sub) (OP6 (AEXT) | (N32_AEXT_ ## sub << 20))
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/hardware/google/interfaces/media/c2/1.0/ |
IComponentStore.hal | 91 AUDIO,
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/external/webrtc/webrtc/ |
call.h | 32 AUDIO, 87 // Audio Processing Module to be used in this call.
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/frameworks/av/media/libstagefright/codecs/common/include/ |
voIndex.h | 71 _MAKE_SOURCE_ID (0x050000, AUDIO) 105 // define audio codec modules 134 _MAKE_SINK_ID (0x020000, AUDIO)
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/external/autotest/client/common_lib/cros/cfm/metrics/ |
media_info_metrics_extractor.py | 22 AUDIO = 1 101 is to sum the received audio bytes for all streams for one logged
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media_metrics_collector.py | 112 media_type=MEDIA_TYPE.AUDIO,
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
rtc_event_log_source.cc | 41 if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
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/external/webrtc/webrtc/call/ |
rtc_event_log2rtp_dump.cc | 33 "Excludes audio packets from the converted RTPdump file."); 128 if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO) 168 if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
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rtc_event_log_unittest.cc | 62 case rtclog::MediaType::AUDIO: 63 return MediaType::AUDIO; 111 << "audio receiver config"; 118 << "audio sender config"; 307 RTPSender rtp_sender(false, // bool audio 483 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, 488 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, 529 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, 536 rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO, 645 log_dumper->LogRtpHeader(false, MediaType::AUDIO, old_rtp_packet.data() [all...] |
rtc_event_log.cc | 151 case MediaType::AUDIO: 152 return rtclog::MediaType::AUDIO;
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/hardware/interfaces/contexthub/1.0/ |
types.hal | 68 AUDIO = 0x300, 69 // Reserving this space for variants on Audio
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/frameworks/av/media/libstagefright/mpeg2ts/ |
ATSParser.h | 75 AUDIO = 1,
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/frameworks/av/media/extractors/mpeg2/ |
MPEG2TSExtractor.cpp | 70 // If there are both audio and video streams, only the video stream 143 // The seek reference track (video if present; audio otherwise) performs 232 sp<AnotherPacketSource> impl = mParser->getSource(ATSParser::AUDIO); 262 // Wait only for 2 seconds to detect audio/video streams. 272 : mParser->getSource(ATSParser::AUDIO); 455 for (auto t: {ATSParser::VIDEO, ATSParser::AUDIO}) {
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/hardware/google/av/media/sfplugin/ |
CCodecConfig.cpp | 412 .limitTo(D::AUDIO)); // read back to both formats 414 .limitTo(D::AUDIO & D::CODED)); 417 .limitTo(D::AUDIO)); // read back to both port formats 419 .limitTo(D::AUDIO & D::CODED)); 422 .limitTo(D::AUDIO)); 425 .limitTo(D::AUDIO & D::CODED) 450 constexpr char KEY_AUDIO_SESSION_ID[] = "audio-session-id"; 519 if (mediaType.startsWith("audio/")) { [all...] |
CCodecConfig.h | 51 * and-ing these groups: e.g. (DECODER | ENCODER) & AUDIO. 58 IS_AUDIO = (1 << 2), ///< for audio codecs 79 AUDIO = ~(IS_IMAGE | IS_VIDEO | OTHER_DOMAIN),
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/frameworks/av/media/libmediaplayerservice/nuplayer/ |
StreamingSource.cpp | 191 sp<AnotherPacketSource> audioTrack = getSource(true /*audio*/); 192 sp<AnotherPacketSource> videoTrack = getSource(false /*audio*/); 200 ALOGV("audio track doesn't have enough data yet. (%.2f secs buffered)", 222 sp<AnotherPacketSource> NuPlayer::StreamingSource::getSource(bool audio) { 228 audio ? ATSParser::AUDIO : ATSParser::VIDEO); 233 sp<AMessage> NuPlayer::StreamingSource::getFormat(bool audio) { 234 sp<AnotherPacketSource> source = getSource(audio); 255 bool audio, sp<ABuffer> *accessUnit) { 256 sp<AnotherPacketSource> source = getSource(audio); [all...] |
/frameworks/base/core/java/android/bluetooth/ |
BluetoothClass.java | 30 * headset, and whether it's capable of services such as audio or telephony. 122 public static final int AUDIO = 0x200000; 132 * BluetoothClass.Service#AUDIO}.
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/external/webrtc/talk/media/webrtc/ |
fakewebrtccall.cc | 36 #include "webrtc/audio/audio_sink.h" 404 media_type == webrtc::MediaType::AUDIO) {
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