/frameworks/base/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/audio/ |
H A D | SimTonesTest.java | 17 package com.android.mediaframeworktest.functional.audio;
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H A D | MediaBassBoostTest.java | 17 package com.android.mediaframeworktest.functional.audio;
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H A D | MediaPresetReverbTest.java | 17 package com.android.mediaframeworktest.functional.audio; 210 // creating a volume controller on output mix ensures that ro.audio.silent mutes 211 // audio after the effects and not before 281 // creating a volume controller on output mix ensures that ro.audio.silent mutes 282 // audio after the effects and not before
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H A D | MediaVirtualizerTest.java | 17 package com.android.mediaframeworktest.functional.audio;
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H A D | MediaAudioManagerTest.java | 17 package com.android.mediaframeworktest.functional.audio; 155 // the audio focus request is async, so wait a bit to verify it had the expected effect 173 //Test case 1: test audio focus listener loses audio focus: 184 //Test case 2: test audio focus listener loses audio focus: 196 //Test case 3: test audio focus listener loses audio focus: 208 //Test case 4: test audio focus registering and use over 3000 iterations 218 assertTrue("audio focu [all...] |
H A D | MediaEnvReverbTest.java | 17 package com.android.mediaframeworktest.functional.audio; 365 // creating a volume controller on output mix ensures that ro.audio.silent mutes 366 // audio after the effects and not before 438 // creating a volume controller on output mix ensures that ro.audio.silent mutes 439 // audio after the effects and not before
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H A D | MediaEqualizerTest.java | 17 package com.android.mediaframeworktest.functional.audio;
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H A D | MediaVisualizerTest.java | 17 package com.android.mediaframeworktest.functional.audio; 242 // creating a volume controller on output mix ensures that ro.audio.silent mutes 243 // audio after the effects and not before 321 // creating a volume controller on output mix ensures that ro.audio.silent mutes 322 // audio after the effects and not before 454 // creating a volume controller on output mix ensures that ro.audio.silent mutes 455 // audio after the effects and not before
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H A D | MediaAudioEffectTest.java | 17 package com.android.mediaframeworktest.functional.audio; 276 //Test case 1.4: test contructor on mediaPlayer audio session
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H A D | MediaAudioTrackTest.java | 17 package com.android.mediaframeworktest.functional.audio; 758 // count (given the audio track properties), and add 77.
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/frameworks/base/services/core/java/com/android/server/audio/ |
H A D | PlayerFocusEnforcer.java | 17 package com.android.server.audio;
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H A D | AudioEventLogger.java | 17 package com.android.server.audio;
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H A D | RotationHelper.java | 17 package com.android.server.audio; 29 * to the audio HALs through AudioSystem. 36 * query the display rotation so audio stays in sync with video/dialogs. This is 82 // use display rotation so audio stays in sync with video/dialogs
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H A D | AudioServiceEvents.java | 17 package com.android.server.audio; 22 import com.android.server.audio.AudioService.WiredDeviceConnectionState;
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H A D | RecordingActivityMonitor.java | 17 package com.android.server.audio;
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H A D | FocusRequester.java | 17 package com.android.server.audio; 29 import com.android.server.audio.MediaFocusControl.AudioFocusDeathHandler; 35 * Class to handle all the information about a user of audio focus. The lifecycle of each 36 * instance is managed by android.media.MediaFocusControl, from its addition to the audio focus 55 * the audio focus gain request that caused the addition of this object in the focus stack. 64 * the audio focus loss received my mFocusDispatcher, is AudioManager.AUDIOFOCUS_NONE if 73 * the audio attributes associated with the focus request 262 * For a given audio focus gain request, return the audio focus loss type that will result 265 * @return the audio focu [all...] |
H A D | MediaFocusControl.java | 17 package com.android.server.audio; 134 * Arbitrary maximum size of audio focus stack to prevent apps OOM'ing this process. 142 * Discard the current audio focus owner. 143 * Notify top of audio focus stack that it lost focus (regardless of possibility to reassign 177 // going through the audio focus stack to signal new focus, traversing order doesn't 196 * Display in the log the current entries in the audio focus stack 266 // is the owner of the audio focus part of the client to remove? 338 * Push the focus requester onto the audio focus stack at the first position immediately 366 * Inner class to monitor audio focus client deaths, and remove them from the audio focu [all...] |
/frameworks/base/core/java/android/speech/tts/ |
H A D | UtteranceProgressListener.java | 18 * be soon before audio is played back in the case of a {@link TextToSpeech#speak} 28 * All audio will have been played back by this point for audible output, and all 79 * Called when the TTS engine begins to synthesize the audio for a request. 87 * This is called when the TTS engine starts synthesizing audio for the request. If an 88 * application wishes to know when the audio is about to start playing, {#onStart(String)} 93 * @param sampleRateInHz Sample rate in hertz of the generated audio. 94 * @param audioFormat Audio format of the generated audio. Should be one of 103 * This is called when a chunk of audio is ready for consumption. 106 * The audio parameter is a copy of what will be synthesized to the speakers (when synthesis was 108 * {@link TextToSpeech#synthesizeToFile}). The audio byte 121 onAudioAvailable(String utteranceId, byte[] audio) argument [all...] |
/frameworks/av/media/libmediaplayer2/nuplayer2/ |
H A D | HTTPLiveSource2.cpp | 129 sp<MetaData> NuPlayer2::HTTPLiveSource2::getFormatMeta(bool audio) { argument 133 audio ? LiveSession::STREAMTYPE_AUDIO 141 sp<AMessage> NuPlayer2::HTTPLiveSource2::getFormat(bool audio) { argument 146 audio ? LiveSession::STREAMTYPE_AUDIO 169 bool audio, sp<ABuffer> *accessUnit) { 171 audio ? LiveSession::STREAMTYPE_AUDIO 339 sp<AMessage> format = getFormat(false /* audio */); 382 bool audio = changedMask & LiveSession::STREAMTYPE_AUDIO; local 390 notify->setInt32("audio", audio); 168 dequeueAccessUnit( bool audio, sp<ABuffer> *accessUnit) argument [all...] |
H A D | NuPlayer2.h | 263 inline const sp<DecoderBase> &getDecoder(bool audio) { argument 264 return audio ? mAudioDecoder : mVideoDecoder; 288 bool audio, sp<DecoderBase> *decoder, bool checkAudioModeChange = true); 297 void handleFlushComplete(bool audio, bool isDecoder); 306 void flushDecoder(bool audio, bool needShutdown); 319 void performDecoderFlush(FlushCommand audio, FlushCommand video); 330 bool audio, bool video, const sp<AMessage> &reply);
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H A D | RTSPSource2.cpp | 156 sp<MetaData> NuPlayer2::RTSPSource2::getFormatMeta(bool audio) { argument 157 sp<AnotherPacketSource> source = getSource(audio); 185 ALOGV("audio track doesn't have enough data yet. (%.2f secs buffered)", 203 bool audio, sp<ABuffer> *accessUnit) { 208 sp<AnotherPacketSource> source = getSource(audio); 219 if (sourceReachedEOS(!audio)) { 228 int64_t eosTimeout = audio ? mEOSTimeoutAudio : mEOSTimeoutVideo; 230 setEOSTimeout(audio, ALooper::GetNowUs()); 232 setEOSTimeout(audio, 0); 238 if (!sourceNearEOS(!audio)) { 202 dequeueAccessUnit( bool audio, sp<ABuffer> *accessUnit) argument 254 getSource(bool audio) argument 265 setEOSTimeout(bool audio, int64_t timeout) argument 417 const bool audio = true; local 431 sourceReachedEOS(bool audio) argument 439 sourceNearEOS(bool audio) argument 461 const bool audio = true; local [all...] |
/frameworks/av/media/libmediaplayerservice/nuplayer/ |
H A D | HTTPLiveSource.cpp | 128 sp<MetaData> NuPlayer::HTTPLiveSource::getFormatMeta(bool audio) { argument 132 audio ? LiveSession::STREAMTYPE_AUDIO 140 sp<AMessage> NuPlayer::HTTPLiveSource::getFormat(bool audio) { argument 145 audio ? LiveSession::STREAMTYPE_AUDIO 168 bool audio, sp<ABuffer> *accessUnit) { 170 audio ? LiveSession::STREAMTYPE_AUDIO 338 sp<AMessage> format = getFormat(false /* audio */); 381 bool audio = changedMask & LiveSession::STREAMTYPE_AUDIO; local 389 notify->setInt32("audio", audio); 167 dequeueAccessUnit( bool audio, sp<ABuffer> *accessUnit) argument [all...] |
H A D | 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) { argument 228 audio ? ATSParser::AUDIO : ATSParser::VIDEO); 233 sp<AMessage> NuPlayer::StreamingSource::getFormat(bool audio) { argument 234 sp<AnotherPacketSource> source = getSource(audio); 255 bool audio, sp<ABuffer> *accessUnit) { 256 sp<AnotherPacketSource> source = getSource(audio); 254 dequeueAccessUnit( bool audio, sp<ABuffer> *accessUnit) argument
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H A D | NuPlayer.h | 270 inline const sp<DecoderBase> &getDecoder(bool audio) { argument 271 return audio ? mAudioDecoder : mVideoDecoder; 289 bool audio, sp<DecoderBase> *decoder, bool checkAudioModeChange = true); 299 void handleFlushComplete(bool audio, bool isDecoder); 310 void flushDecoder(bool audio, bool needShutdown); 323 void performDecoderFlush(FlushCommand audio, FlushCommand video); 333 bool audio, bool video, const sp<AMessage> &reply);
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H A D | RTSPSource.cpp | 158 sp<MetaData> NuPlayer::RTSPSource::getFormatMeta(bool audio) { argument 159 sp<AnotherPacketSource> source = getSource(audio); 187 ALOGV("audio track doesn't have enough data yet. (%.2f secs buffered)", 205 bool audio, sp<ABuffer> *accessUnit) { 210 sp<AnotherPacketSource> source = getSource(audio); 221 if (sourceReachedEOS(!audio)) { 230 int64_t eosTimeout = audio ? mEOSTimeoutAudio : mEOSTimeoutVideo; 232 setEOSTimeout(audio, ALooper::GetNowUs()); 234 setEOSTimeout(audio, 0); 240 if (!sourceNearEOS(!audio)) { 204 dequeueAccessUnit( bool audio, sp<ABuffer> *accessUnit) argument 256 getSource(bool audio) argument 267 setEOSTimeout(bool audio, int64_t timeout) argument 419 const bool audio = true; local 433 sourceReachedEOS(bool audio) argument 441 sourceNearEOS(bool audio) argument 463 const bool audio = true; local [all...] |