/frameworks/base/libs/audioflinger/ |
H A D | AudioHardwareStub.cpp | 46 int format, int channelCount, uint32_t sampleRate, status_t *status) 49 status_t lStatus = out->set(format, channelCount, sampleRate); 60 int inputSource, int format, int channelCount, uint32_t sampleRate, 70 status_t lStatus = in->set(format, channelCount, sampleRate, acoustics); 115 if (rate == 0) rate = sampleRate(); 119 (rate == sampleRate())) 127 usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate()); 142 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); 158 (rate == sampleRate())) 166 usleep(bytes * 1000000 / sizeof(int16_t) / channelCount() / sampleRate()); 45 openOutputStream( int format, int channelCount, uint32_t sampleRate, status_t *status) argument 59 openInputStream( int inputSource, int format, int channelCount, uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument [all...] |
H A D | A2dpAudioInterface.h | 54 uint32_t sampleRate=0, 61 uint32_t sampleRate, 76 uint32_t sampleRate); 77 virtual uint32_t sampleRate() const { return 44100; } function in class:android::A2dpAudioInterface::A2dpAudioStreamOut 82 virtual uint32_t latency() const { return ((1000*bufferSize())/frameSize())/sampleRate() + 200; }
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H A D | AudioHardwareStub.h | 32 virtual status_t set(int format, int channelCount, uint32_t sampleRate); 33 virtual uint32_t sampleRate() const { return 44100; } function in class:android::AudioStreamOutStub 46 virtual status_t set(int format, int channelCount, uint32_t sampleRate, AudioSystem::audio_in_acoustics acoustics); 47 virtual uint32_t sampleRate() const { return 8000; } function in class:android::AudioStreamInStub 77 uint32_t sampleRate=0, 84 uint32_t sampleRate,
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H A D | AudioDumpInterface.h | 36 virtual uint32_t sampleRate() const { return mFinalStream->sampleRate(); } function in class:android::AudioStreamOutDump 61 uint32_t sampleRate=0, 82 uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) 83 { return mFinalInterface->openInputStream(inputSource, format, channelCount, sampleRate, status, acoustics); } 81 openInputStream(int inputSource, int format, int channelCount, uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument
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H A D | A2dpAudioInterface.cpp | 48 int format, int channelCount, uint32_t sampleRate, status_t *status) 50 LOGD("A2dpAudioInterface::openOutputStream %d, %d, %d\n", format, channelCount, sampleRate); 62 if ((err = out->set(format, channelCount, sampleRate)) == NO_ERROR) { 74 int inputSource, int format, int channelCount, uint32_t sampleRate, 150 if (rate == 0) rate = sampleRate(); 155 (rate != sampleRate())) 198 usleep(bytes * 1000000 / frameSize() / sampleRate()); 47 openOutputStream( int format, int channelCount, uint32_t sampleRate, status_t *status) argument 73 openInputStream( int inputSource, int format, int channelCount, uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument
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H A D | AudioHardwareGeneric.h | 44 uint32_t sampleRate); 46 virtual uint32_t sampleRate() const { return 44100; } function in class:android::AudioStreamOutGeneric 72 uint32_t sampleRate, 75 uint32_t sampleRate() const { return 8000; } function in class:android::AudioStreamInGeneric 111 uint32_t sampleRate=0, 118 uint32_t sampleRate,
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H A D | AudioHardwareGeneric.cpp | 66 int format, int channelCount, uint32_t sampleRate, status_t *status) 80 status_t lStatus = out->set(this, mFd, format, channelCount, sampleRate); 97 int inputSource, int format, int channelCount, uint32_t sampleRate, 118 status_t lStatus = in->set(this, mFd, format, channelCount, sampleRate, acoustics); 195 if (rate == 0) rate = sampleRate(); 200 (rate != sampleRate())) 233 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); 265 (rate != sampleRate())) { 302 snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate()); 65 openOutputStream( int format, int channelCount, uint32_t sampleRate, status_t *status) argument 96 openInputStream( int inputSource, int format, int channelCount, uint32_t sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) argument
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H A D | AudioDumpInterface.cpp | 53 int format, int channelCount, uint32_t sampleRate, status_t *status) 55 AudioStreamOut* outFinal = mFinalInterface->openOutputStream(format, channelCount, sampleRate, status); 52 openOutputStream( int format, int channelCount, uint32_t sampleRate, status_t *status) argument
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H A D | AudioResampler.h | 45 int32_t sampleRate, int quality=DEFAULT); 67 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
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H A D | AudioResampler.cpp | 38 AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : argument 39 AudioResampler(bitDepth, inChannelCount, sampleRate), mX0L(0), mX0R(0) { 78 int32_t sampleRate, int quality) { 96 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); 100 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); 104 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); 114 int32_t sampleRate) : 116 mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), 77 create(int bitDepth, int inChannelCount, int32_t sampleRate, int quality) argument 113 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate) argument
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H A D | AudioResamplerCubic.h | 31 AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) : argument 32 AudioResampler(bitDepth, inChannelCount, sampleRate) {
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H A D | AudioHardwareInterface.cpp | 209 size_t AudioHardwareBase::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument 211 if (sampleRate != 8000) { 212 LOGW("getInputBufferSize bad sampling rate: %d", sampleRate);
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H A D | AudioMixer.h | 39 AudioMixer(size_t frameCount, uint32_t sampleRate); 147 uint32_t sampleRate; member in struct:android::AudioMixer::track_t 149 bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
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H A D | AudioFlinger.h | 70 uint32_t sampleRate, 78 virtual uint32_t sampleRate(int output) const; 113 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount); 143 uint32_t sampleRate, 235 uint32_t sampleRate, 268 int sampleRate() const; 308 uint32_t sampleRate, 375 uint32_t sampleRate, 415 uint32_t sampleRate, 448 virtual uint32_t sampleRate() cons [all...] |
H A D | AudioResamplerSinc.h | 32 AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate);
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/frameworks/base/core/java/android/speech/srec/ |
H A D | MicrophoneInputStream.java | 42 * @param sampleRate sample rate of the microphone, typically 11025 or 8000. 43 * @param fifoDepth depth of the real time fifo, measured in sampleRate clock ticks. 46 public MicrophoneInputStream(int sampleRate, int fifoDepth) throws IOException { argument 47 mAudioRecord = AudioRecordNew(sampleRate, fifoDepth); 105 private static native int AudioRecordNew(int sampleRate, int fifoDepth); argument
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H A D | WaveHeader.java | 29 * <li> sampleRate - usually 8000, 11025, 16000, 22050, or 44100 hz. 69 * @param sampleRate typically 8000, 11025, 16000, 22050, or 44100 hz. 73 public WaveHeader(short format, short numChannels, int sampleRate, short bitsPerSample, int numBytes) { argument 75 mSampleRate = sampleRate; 129 * @param sampleRate sample rate, typically 8000, 11025, 16000, 22050, or 44100 hz. 132 public WaveHeader setSampleRate(int sampleRate) { argument 133 mSampleRate = sampleRate; 272 "WaveHeader format=%d numChannels=%d sampleRate=%d bitsPerSample=%d numBytes=%d",
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/frameworks/base/include/media/ |
H A D | IAudioFlinger.h | 47 uint32_t sampleRate, 58 uint32_t sampleRate, 68 virtual uint32_t sampleRate(int output) const = 0; 116 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
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H A D | AudioTrack.h | 119 * sampleRate: Track sampling rate in Hz. 134 uint32_t sampleRate = 0, 153 uint32_t sampleRate = 0, 172 * - BAD_VALUE: invalid parameter (channelCount, format, sampleRate...) 176 uint32_t sampleRate = 0, 248 status_t setSampleRate(int sampleRate);
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/frameworks/base/include/private/media/ |
H A D | AudioTrackShared.h | 57 uint32_t sampleRate; member in struct:android::audio_track_cblk_t
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/frameworks/base/media/libmedia/ |
H A D | AudioRecord.cpp | 54 uint32_t sampleRate, 64 mStatus = set(inputSource, sampleRate, format, channelCount, 86 uint32_t sampleRate, 97 LOGV("set(): sampleRate %d, channelCount %d, frameCount %d",sampleRate, channelCount, frameCount); 111 if (sampleRate == 0) { 112 sampleRate = DEFAULT_SAMPLE_RATE; 132 if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &inputBuffSizeInBytes) 138 LOGE("Recording parameters are not supported: sampleRate %d, channelCount %d, format %d", 139 sampleRate, channelCoun 52 AudioRecord( int inputSource, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames) argument 84 set( int inputSource, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, bool threadCanCallJava) argument [all...] |
H A D | IMediaPlayerService.cpp | 151 uint32_t sampleRate; local 154 sp<IMemory> player = decode(url, &sampleRate, &numChannels, &format); 155 reply->writeInt32(sampleRate); 166 uint32_t sampleRate; local 169 sp<IMemory> player = decode(fd, offset, length, &sampleRate, &numChannels, &format); 170 reply->writeInt32(sampleRate);
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H A D | IAudioFlinger.cpp | 71 uint32_t sampleRate, 83 data.writeInt32(sampleRate); 103 uint32_t sampleRate, 114 data.writeInt32(sampleRate); 127 virtual uint32_t sampleRate(int output) const function in class:android::BpAudioFlinger 324 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) argument 328 data.writeInt32(sampleRate); 370 uint32_t sampleRate = data.readInt32(); local 378 streamType, sampleRate, format, 388 uint32_t sampleRate local 68 createTrack( pid_t pid, int streamType, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, const sp<IMemory>& sharedBuffer, status_t *status) argument 100 openRecord( pid_t pid, int inputSource, uint32_t sampleRate, int format, int channelCount, int frameCount, uint32_t flags, status_t *status) argument 530 uint32_t sampleRate = data.readInt32(); local [all...] |
H A D | AudioSystem.cpp | 71 gOutSamplingRate[output] = (int)gAudioFlinger->sampleRate(output); 300 status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount, argument 304 if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat) 307 gPrevInSamplingRate = sampleRate; 316 gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
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/frameworks/base/media/jni/soundpool/ |
H A D | SoundPool.cpp | 428 uint32_t sampleRate; local 434 p = MediaPlayer::decode(mUrl, &sampleRate, &numChannels, &format); 436 p = MediaPlayer::decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format); 445 LOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d", 446 p->pointer(), p->size(), sampleRate, numChannels); 448 if (sampleRate > kMaxSampleRate) { 449 LOGE("Sample rate (%u) out of range", sampleRate); 464 mSampleRate = sampleRate; 503 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rat local 707 uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5); local [all...] |