/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
H A D | bitenc.h | 35 Word32 sampleRate; member in struct:BITSTREAMENCODER_INIT
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H A D | aacenc_core.h | 40 Word32 sampleRate; /* audio file sample rate */ member in struct:__anon530
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/frameworks/av/media/libstagefright/codecs/common/include/ |
H A D | voAAC.h | 45 int sampleRate; /*! audio file sample rate */ member in struct:__anon649
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/frameworks/av/media/libnbaio/ |
H A D | AudioStreamInSource.cpp | 47 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local 50 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
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H A D | AudioStreamOutSink.cpp | 44 uint32_t sampleRate = mStream->common.get_sample_rate(&mStream->common); local 47 mFormat = Format_from_SR_C(sampleRate, popcount(channelMask));
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H A D | NBAIO.cpp | 96 NBAIO_Format Format_from_SR_C(unsigned sampleRate, unsigned channelCount) argument 99 switch (sampleRate) {
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/frameworks/opt/net/voip/src/jni/rtp/ |
H A D | GsmCodec.cpp | 42 int set(int sampleRate, const char *fmtp) { argument 43 return (sampleRate == 8000 && mEncode && mDecode) ? 160 : -1;
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H A D | G711Codec.cpp | 37 int set(int sampleRate, const char *fmtp) { argument 38 mSampleCount = sampleRate / 50; 88 int set(int sampleRate, const char *fmtp) { argument 89 mSampleCount = sampleRate / 50;
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H A D | AmrCodec.cpp | 53 int set(int sampleRate, const char *fmtp); 67 int AmrCodec::set(int sampleRate, const char *fmtp) argument 97 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1; 211 int set(int sampleRate, const char *fmtp) { argument 212 return (sampleRate == 8000 && mEncoder && mDecoder) ? 160 : -1;
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/frameworks/av/cmds/stagefright/ |
H A D | SineSource.cpp | 12 SineSource::SineSource(int32_t sampleRate, int32_t numChannels) argument 14 mSampleRate(sampleRate),
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H A D | SimplePlayer.cpp | 583 int32_t sampleRate; local 585 CHECK(format->findInt32("sample-rate", &sampleRate)); 589 sampleRate,
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/frameworks/av/media/libeffects/testlibs/ |
H A D | AudioPeakingFilter.cpp | 44 AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate) argument 45 : mBiquad(nChannels, sampleRate) { 46 configure(nChannels, sampleRate); 50 void AudioPeakingFilter::configure(int nChannels, int sampleRate) { argument 51 mNiquistFreq = sampleRate * 500; 53 mBiquad.configure(nChannels, sampleRate);
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H A D | AudioShelvingFilter.cpp | 50 int sampleRate) 52 mBiquad(nChannels, sampleRate) { 53 configure(nChannels, sampleRate); 56 void AudioShelvingFilter::configure(int nChannels, int sampleRate) { argument 57 mNiquistFreq = sampleRate * 500; 59 mBiquad.configure(nChannels, sampleRate); 49 AudioShelvingFilter(ShelfType type, int nChannels, int sampleRate) argument
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H A D | AudioBiquadFilter.cpp | 28 AudioBiquadFilter::AudioBiquadFilter(int nChannels, int sampleRate) { argument 29 configure(nChannels, sampleRate); 33 void AudioBiquadFilter::configure(int nChannels, int sampleRate) { argument 35 assert(sampleRate > 0); 39 / sampleRate;
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H A D | AudioEqualizer.cpp | 39 int nChannels, int sampleRate, 43 "sampleRate=%d, nPresets=%d)", 44 pMem, nBands, nChannels, sampleRate, nPresets); 54 return new (pMem) AudioEqualizer(pMem, nBands, nChannels, sampleRate, 58 void AudioEqualizer::configure(int nChannels, int sampleRate) { argument 59 ALOGV("AudioEqualizer::configure(nChannels=%d, sampleRate=%d)", nChannels, 60 sampleRate); 61 mpLowShelf->configure(nChannels, sampleRate); 63 mpPeakingFilters[i].configure(nChannels, sampleRate); 65 mpHighShelf->configure(nChannels, sampleRate); 38 CreateInstance(void * pMem, int nBands, int nChannels, int sampleRate, const PresetConfig * presets, int nPresets) argument 287 AudioEqualizer(void * pMem, int nBands, int nChannels, int sampleRate, bool ownMem, const PresetConfig * presets, int nPresets) argument [all...] |
/frameworks/av/media/libstagefright/rtsp/ |
H A D | ARawAudioAssembler.cpp | 134 int32_t sampleRate, numChannels; local 136 desc, &sampleRate, &numChannels); 138 format->setInt32(kKeySampleRate, sampleRate);
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/frameworks/av/services/audioflinger/ |
H A D | AudioResamplerCubic.h | 31 AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) : argument 32 AudioResampler(bitDepth, inChannelCount, sampleRate, MED_QUALITY) {
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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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/frameworks/av/media/libstagefright/ |
H A D | AMRWriter.cpp | 91 int32_t sampleRate; local 94 CHECK(meta->findInt32(kKeySampleRate, &sampleRate)); 95 CHECK_EQ(sampleRate, (isWide ? 16000 : 8000));
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H A D | VBRISeeker.cpp | 49 int sampleRate; local 50 if (!GetMPEGAudioFrameSize(tmp, &frameSize, &sampleRate)) { 70 numFrames * 1000000ll * (sampleRate >= 32000 ? 1152 : 576) / sampleRate;
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H A D | AACWriter.cpp | 210 static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) { argument 220 if (sampleRate == kSampleRateTable[index]) { 222 sampleRate, index); 228 ALOGE("Sampling rate %d bps is not supported", sampleRate);
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H A D | AudioSource.cpp | 51 audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount) 54 mSampleRate(sampleRate), 58 ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount); 63 sampleRate, 78 inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT, 50 AudioSource( audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount) argument
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/frameworks/base/core/java/android/speech/tts/ |
H A D | SynthesisPlaybackQueueItem.java | 66 SynthesisPlaybackQueueItem(int streamType, int sampleRate, argument 78 mAudioTrack = new BlockingAudioTrack(streamType, sampleRate, audioFormat,
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/frameworks/av/libvideoeditor/vss/src/ |
H A D | VideoEditorResampler.cpp | 79 M4OSA_Int32 sampleRate, M4OSA_Int32 quality) { 83 bitDepth, inChannelCount, sampleRate); 90 context->outSamplingRate = sampleRate; 78 LVAudioResamplerCreate(M4OSA_Int32 bitDepth, M4OSA_Int32 inChannelCount, M4OSA_Int32 sampleRate, M4OSA_Int32 quality) argument
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/frameworks/av/media/libmedia/ |
H A D | IMediaPlayerService.cpp | 208 uint32_t sampleRate; local 211 sp<IMemory> player = decode(url, &sampleRate, &numChannels, &format); 212 reply->writeInt32(sampleRate); 223 uint32_t sampleRate; local 226 sp<IMemory> player = decode(fd, offset, length, &sampleRate, &numChannels, &format); 227 reply->writeInt32(sampleRate);
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