/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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H A D | APacketSource.cpp | 473 int32_t sampleRate, numChannels; local 475 desc.c_str(), &sampleRate, &numChannels); 477 mFormat->setInt32(kKeySampleRate, sampleRate); 489 int32_t sampleRate, numChannels; local 491 desc.c_str(), &sampleRate, &numChannels); 493 mFormat->setInt32(kKeySampleRate, sampleRate); 496 if (sampleRate != 8000 || numChannels != 1) { 502 int32_t sampleRate, numChannels; local 504 desc.c_str(), &sampleRate, &numChannels); 506 mFormat->setInt32(kKeySampleRate, sampleRate); 553 int32_t sampleRate, numChannels; local [all...] |
/frameworks/av/services/audioflinger/ |
H A D | AudioResampler.cpp | 42 AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : argument 43 AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { 138 int32_t sampleRate, src_quality quality) { 191 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); 196 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); 201 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); 205 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality); 215 int32_t sampleRate, src_quality quality) : 217 mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputInde 137 create(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality) argument 214 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality) argument [all...] |
H A D | AudioResampler.h | 46 int32_t sampleRate, src_quality quality=DEFAULT_QUALITY); 77 AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
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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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H A D | AudioResamplerSinc.h | 37 AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate,
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/frameworks/av/include/media/ |
H A D | AudioRecord.h | 102 uint32_t sampleRate, 119 * sampleRate: Track sampling rate in Hz. 134 uint32_t sampleRate = 0, 154 * - BAD_VALUE: invalid parameter (channels, format, sampleRate...) 159 uint32_t sampleRate = 0, 344 status_t openRecord_l(uint32_t sampleRate,
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H A D | IAudioFlinger.h | 61 uint32_t sampleRate, 75 uint32_t sampleRate, 87 virtual uint32_t sampleRate(audio_io_handle_t output) const = 0; 132 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
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/frameworks/av/media/libstagefright/ |
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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/frameworks/av/media/libstagefright/codecs/aacenc/SampleCode/ |
H A D | AAC_E_SAMPLES.c | 53 // bitRate/nChannels < sampleRate*6 57 param->sampleRate = 44100; 84 param->sampleRate = atoi(*argv); 116 if(param->sampleRate%8000 == 0) 118 param->bitRate = 640*param->nChannels*param->sampleRate/scale;
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/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
H A D | aacenc_core.h | 40 Word32 sampleRate; /* audio file sample rate */ member in struct:__anon515
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H A D | block_switch.h | 63 Word32 sampleRate,
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H A D | qc_main.h | 62 Word32 sampleRate);
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H A D | psy_configuration.h | 91 Word32 GetSRIndex(Word32 sampleRate);
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/frameworks/base/core/java/android/speech/srec/ |
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/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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/frameworks/base/voip/jni/rtp/ |
H A D | AudioGroup.cpp | 100 AudioCodec *codec, int sampleRate, int sampleCount, 104 bool mix(int32_t *output, int head, int tail, int sampleRate); 166 AudioCodec *codec, int sampleRate, int sampleCount, 178 mSampleRate = sampleRate / 1000; 234 bool AudioStream::mix(int32_t *output, int head, int tail, int sampleRate) argument 253 if (sampleRate == mSampleRate) { 476 bool set(int sampleRate, int sampleCount); 572 bool AudioGroup::set(int sampleRate, int sampleCount) argument 580 mSampleRate = sampleRate; 594 sampleRate, sampleCoun 165 set(int mode, int socket, sockaddr_storage *remote, AudioCodec *codec, int sampleRate, int sampleCount, int codecType, int dtmfType) argument 783 int sampleRate = mGroup->mSampleRate; local 966 int sampleRate = -1; local [all...] |
/frameworks/av/media/libstagefright/codecs/aacdec/ |
H A D | SoftAAC2.cpp | 185 aacParams->nSampleRate = mStreamInfo->sampleRate; 218 pcmParams->nSamplingRate = mStreamInfo->sampleRate; 336 if (mStreamInfo->sampleRate && mStreamInfo->numChannels) { 339 mStreamInfo->sampleRate, 467 int prevSampleRate = mStreamInfo->sampleRate; 502 if (mStreamInfo->sampleRate != prevSampleRate || 506 mStreamInfo->sampleRate, 519 } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) { 563 + (mNumSamplesOutput * 1000000ll) / mStreamInfo->sampleRate;
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/frameworks/av/media/libstagefright/codecs/aacenc/src/ |
H A D | qc_main.c | 61 Word32 sampleRate, 69 quot = result / sampleRate; 73 result -= quot * sampleRate; 90 Word32 sampleRate, 99 sampleRate, 106 *paddingRest = *paddingRest + sampleRate; 546 Word32 sampleRate) /* output sampling rate */ 555 sampleRate, 560 sampleRate, 60 calcFrameLen(Word32 bitRate, Word32 sampleRate, FRAME_LEN_RESULT_MODE mode) argument 89 framePadding(Word32 bitRate, Word32 sampleRate, Word32 *paddingRest) argument 544 AdjustBitrate(QC_STATE *hQC, Word32 bitRate, Word32 sampleRate) argument
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/frameworks/av/media/libeffects/testlibs/ |
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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/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 | AudioTrack.cpp | 55 uint32_t sampleRate) 85 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 86 afFrameCount * minBufCount * sampleRate / afSampleRate; 104 uint32_t sampleRate, 118 mStatus = set(streamType, sampleRate, format, channelMask, 126 uint32_t sampleRate, 139 mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, 147 uint32_t sampleRate, 161 mStatus = set(streamType, sampleRate, format, channelMask, 188 uint32_t sampleRate, 52 getMinFrameCount( int* frameCount, audio_stream_type_t streamType, uint32_t sampleRate) argument 102 AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_output_flags_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument 124 AudioTrack( int streamType, uint32_t sampleRate, int format, int channelMask, int frameCount, uint32_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument 145 AudioTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, const sp<IMemory>& sharedBuffer, audio_output_flags_t flags, callback_t cbf, void* user, int notificationFrames, int sessionId) argument 186 set( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_output_flags_t flags, callback_t cbf, void* user, int notificationFrames, const sp<IMemory>& sharedBuffer, bool threadCanCallJava, int sessionId) argument 749 createTrack_l( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, int frameCount, audio_output_flags_t flags, const sp<IMemory>& sharedBuffer, audio_io_handle_t output) argument [all...] |
H A D | IMediaPlayerService.cpp | 182 uint32_t sampleRate; local 185 sp<IMemory> player = decode(url, &sampleRate, &numChannels, &format); 186 reply->writeInt32(sampleRate); 197 uint32_t sampleRate; local 200 sp<IMemory> player = decode(fd, offset, length, &sampleRate, &numChannels, &format); 201 reply->writeInt32(sampleRate);
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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/base/media/java/android/media/ |
H A D | MediaFormat.java | 261 * @param sampleRate The sampling rate of the content. 266 int sampleRate, 270 format.setInteger(KEY_SAMPLE_RATE, sampleRate); 264 createAudioFormat( String mime, int sampleRate, int channelCount) argument
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