| /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)
|
| CD.py | 12 AUDIO = 0
|
| /external/python/cpython2/Lib/plat-irix6/ |
| CD.py | 12 AUDIO = 0
|
| /platform_testing/libraries/app-helpers/handheld/src/android/platform/test/helpers/handheld/ |
| AbstractDownloadsHelper.java | 24 AUDIO,
72 * Setup expectation: Audio is playing
74 * This method will wait for the audio to stop playing or until timeoutInSeconds occur,
77 * @param timeoutInSeconds - timeout value in seconds the test will wait for audio to end
|
| /toolchain/binutils/binutils-2.25/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 | 134 /* for audio-extension. */
295 #define AUDIO(sub) (OP6 (AEXT) | (N32_AEXT_ ## sub << 20))
|
| frv-desc.c | 273 { "AUDIO", &bool_attr[0], &bool_attr[0] },
[all...] |
| /external/webrtc/webrtc/ |
| call.h | 32 AUDIO,
87 // Audio Processing Module to be used in this call.
|
| /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)
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| rtc_event_log_source.cc | 41 if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
|
| /frameworks/av/media/libstagefright/include/ |
| AVIExtractor.h | 65 AUDIO,
|
| /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)
|
| 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;
|
| call_perf_tests.cc | 162 // During the first couple of seconds audio and video can falsely be
207 media_type == MediaType::AUDIO);
338 << "Timed out while waiting for audio and video to be synchronized.";
|
| /hardware/interfaces/contexthub/1.0/ |
| types.hal | 68 AUDIO = 0x300,
69 // Reserving this space for variants on Audio
|
| /frameworks/av/media/libstagefright/mpeg2ts/ |
| ATSParser.h | 75 AUDIO = 1,
|
| MPEG2TSExtractor.cpp | 69 // If there are both audio and video streams, only the video stream
137 // The seek reference track (video if present; audio otherwise) performs
225 ATSParser::AUDIO).get();
255 // Wait only for 2 seconds to detect audio/video streams.
267 ATSParser::AUDIO).get();
451 for (auto t: {ATSParser::VIDEO, ATSParser::AUDIO}) {
|
| ATSParser.cpp | 633 // (eg. video is read first and starts at 0, but audio starts at 0xfffffff0)
657 if (type == AUDIO && stream->isAudio()) {
[all...] |
| /frameworks/av/media/libmediaplayerservice/nuplayer/ |
| StreamingSource.cpp | 195 sp<AnotherPacketSource> audioTrack = getSource(true /*audio*/);
196 sp<AnotherPacketSource> videoTrack = getSource(false /*audio*/);
204 ALOGV("audio track doesn't have enough data yet. (%.2f secs buffered)",
226 sp<AnotherPacketSource> NuPlayer::StreamingSource::getSource(bool audio) {
232 audio ? ATSParser::AUDIO : ATSParser::VIDEO);
237 sp<AMessage> NuPlayer::StreamingSource::getFormat(bool audio) {
238 sp<AnotherPacketSource> source = getSource(audio);
259 bool audio, sp<ABuffer> *accessUnit) {
260 sp<AnotherPacketSource> source = getSource(audio);
[all...] |
| /frameworks/base/core/java/android/bluetooth/ |
| BluetoothClass.java | 26 * headset, and whether it's capable of services such as audio or telephony.
114 public static final int AUDIO = 0x200000;
124 * BluetoothClass.Service#AUDIO}.
|
| /external/webrtc/talk/media/webrtc/ |
| fakewebrtccall.cc | 36 #include "webrtc/audio/audio_sink.h"
404 media_type == webrtc::MediaType::AUDIO) {
|
| /test/vts/compilation_tools/vtsc/ |
| VtsCompilerUtils.cpp | 67 case AUDIO:
68 return "audio";
|
| /frameworks/av/media/libstagefright/ |
| AVIExtractor.cpp | 638 if (mime && strncasecmp(mime, "audio/", 6)) {
642 kind = Track::AUDIO;
709 ALOGW("Unsupported audio format = 0x%04x", format);
737 case Track::AUDIO:
[all...] |
