| /external/libxaac/decoder/ |
| ixheaacd_aacdec.h | 37 ia_aac_dec_channel_info_struct *pstr_aac_dec_ch_info[CHANNELS];
39 ia_aac_dec_channel_info *ptr_aac_dec_static_channel_info[CHANNELS];
41 ia_aac_dec_overlap_info *pstr_aac_dec_overlap_info[CHANNELS];
48 WORD16 channels; member in struct:ia_aac_decoder_struct
56 ia_aac_dec_overlap_info str_aac_dec_overlap_info[CHANNELS];
61 ia_aac_dec_channel_info *ptr_aac_dec_static_channel_info[CHANNELS];
62 WORD16 *ltp_buf[CHANNELS];
82 WORD32 channels);
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| ixheaacd_defines.h | 41 #define CHANNELS 2
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| ixheaacd_channel.h | 51 ia_aac_dec_channel_info_struct *ptr_aac_dec_channel_info[CHANNELS]);
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| ixheaacd_aacdecoder.c | 127 ia_aac_sfb_code_book_struct *ptr_aac_sfb_code_book_data[CHANNELS];
773 aac_dec_handle->channels = num_ch;
792 pstr_drc_dec, aac_dec_handle->channels,
795 for (ch = 0; ch < aac_dec_handle->channels; ch++) {
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| ixheaacd_channel.c | 664 ia_aac_dec_channel_info_struct *ptr_aac_dec_channel_info[CHANNELS]) {
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| /external/libopus/doc/ |
| trivial_example.c | 41 #define CHANNELS 2
54 opus_int16 in[FRAME_SIZE*CHANNELS];
55 opus_int16 out[MAX_FRAME_SIZE*CHANNELS];
71 encoder = opus_encoder_create(SAMPLE_RATE, CHANNELS, APPLICATION, &err);
97 decoder = opus_decoder_create(SAMPLE_RATE, CHANNELS, &err);
114 unsigned char pcm_bytes[MAX_FRAME_SIZE*CHANNELS*2];
118 fread(pcm_bytes, sizeof(short)*CHANNELS, FRAME_SIZE, fin);
122 for (i=0;i<CHANNELS*FRAME_SIZE;i++)
146 for(i=0;i<CHANNELS*frame_size;i++)
152 fwrite(pcm_bytes, sizeof(short), frame_size*CHANNELS, fout)
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| /frameworks/av/media/libaudioprocessing/ |
| AudioResamplerFirProcess.h | 79 template<int CHANNELS, typename TO>
80 class Accumulator : public Accumulator<CHANNELS-1, TO> // recursive
85 Accumulator<CHANNELS-1, TO>::clear();
90 Accumulator<CHANNELS-1, TO>::acc(coef, data);
94 Accumulator<CHANNELS-1, TO>::volume(out, gain);
177 template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO,
189 static_assert(CHANNELS > 0, "CHANNELS must be > 0");
191 if (CHANNELS > 2) {
192 // TO accum[CHANNELS];
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| AudioResamplerSinc.h | 49 template<int CHANNELS>
53 template<int CHANNELS>
57 template<int CHANNELS>
63 template<int CHANNELS>
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| AudioResamplerSinc.cpp | 293 template<int CHANNELS>
298 const size_t headOffset = c.halfNumCoefs*CHANNELS;
319 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
322 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
328 read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
337 for (size_t i=0 ; i<CHANNELS ; i++) {
338 head[i] = in[inputIndex*CHANNELS + i];
343 filterCoefficient<CHANNELS>(&out[outputIndex], phaseFraction, impulse, vRL);
353 read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
368 return outputIndex / CHANNELS;
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| AudioResamplerFirProcessNeon.h | 73 template <int CHANNELS, int STRIDE, bool FIXED>
86 static_assert(CHANNELS == 1 || CHANNELS == 2, "CHANNELS must be 1 or 2");
88 sP -= CHANNELS*((STRIDE>>1)-1);
103 if (CHANNELS == 2) {
127 switch (CHANNELS) {
168 if (CHANNELS == 1) {
171 } else if (CHANNELS == 2) {
181 template <int CHANNELS, int STRIDE, bool FIXED
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| AudioResamplerDyn.h | 113 void resize(int CHANNELS, int halfNumCoefs);
124 template<int CHANNELS>
128 template<int CHANNELS>
138 // in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
150 template<int CHANNELS, bool LOCKED, int STRIDE>
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| AudioResamplerDyn.cpp | 86 void AudioResamplerDyn<TC, TI, TO>::InBuffer::resize(int CHANNELS, int halfNumCoefs)
89 size_t stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength;
94 && mRingFull-mState == (ssize_t) (mStateCount-halfNumCoefs*CHANNELS)) {
108 TI* srcLo = mImpulse - halfNumCoefs*CHANNELS;
109 TI* srcHi = mImpulse + halfNumCoefs*CHANNELS;
126 mImpulse = state + halfNumCoefs*CHANNELS; // actually one sample greater than needed
127 mRingFull = state + mStateCount - halfNumCoefs*CHANNELS;
132 template<int CHANNELS>
136 TI* head = impulse + halfNumCoefs*CHANNELS;
137 for (size_t i=0 ; i<CHANNELS ; i++)
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| AudioResamplerFirProcessSSE.h | 37 template <int CHANNELS, int STRIDE, bool FIXED>
50 static_assert(CHANNELS == 1 || CHANNELS == 2, "CHANNELS must be 1 or 2");
52 sP -= CHANNELS*(4-1); // adjust sP for a loop iteration of four
61 if (CHANNELS == 2) {
89 switch (CHANNELS) {
138 if (CHANNELS == 1) {
142 } else if (CHANNELS == 2) {
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| /external/libopus/tests/ |
| test_opus_padding.c | 39 #define CHANNELS 2
49 opus_int16 *out = malloc(FRAMESIZE*CHANNELS*sizeof(*out));
61 decoder = opus_decoder_create(48000, CHANNELS, &error);
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| /frameworks/av/services/audioflinger/ |
| BufLog.h | 36 * BUFLOG(buff_id, buff_tag, format, channels, sampling_rate, max_bytes, buff_pointer, buff_size);
41 * channels: int Channel Count
52 * int channels = audio_channel_count_from_out_mask(mConfig.outputCfg.channels);
55 * int frameSize = audio_bytes_per_sample((audio_format_t)format) * channels;
58 * BUFLOG(11, "loudnes_enhancer_out", format, channels, samplingRate, maxBytes,
81 #define __BUFLOG(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, BUF, SIZE) \
82 BufLogSingleton::instance()->write(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, \
85 #define BUFLOG(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, BUF, SIZE) \
88 #define BUFLOG(STREAMID, TAG, FORMAT, CHANNELS, SAMPLINGRATE, MAXBYTES, BUF, SIZE)
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| /external/tensorflow/tensorflow/contrib/layers/python/layers/ |
| rev_block_lib_test.py | 39 CHANNELS = 8
46 return core_layers.dense(x, self.CHANNELS // 2, use_bias=True)
49 return core_layers.dense(x, self.CHANNELS // 2, use_bias=True)
52 [self.BATCH_SIZE, self.CHANNELS], dtype=dtypes.float32)
69 return core_layers.dense(x, self.CHANNELS // 2, use_bias=True)
72 return core_layers.dense(x, self.CHANNELS // 2, use_bias=True)
75 [self.BATCH_SIZE, self.CHANNELS], dtype=dtypes.float32)
100 return core_layers.dense(x, self.CHANNELS // 2, use_bias=True)
105 return core_layers.dense(x, self.CHANNELS // 2, use_bias=True)
115 [self.BATCH_SIZE, self.CHANNELS], dtype=dtypes.float32
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| /external/tensorflow/tensorflow/core/kernels/ |
| resize_bicubic_op_test.cc | 115 const int channels = images.dimension(3); local
118 ASSERT_EQ(channels, output.dimension(3));
138 for (int64 c = 0; c < channels; ++c) {
160 const int target_width, int channels) {
162 << channels << " to " << target_height << "x" << target_width
163 << "x" << channels; local
165 TensorShape({batch_size, in_height, in_width, channels}));
173 TensorShape({batch_size, target_height, target_width, channels})));
185 void RunManyRandomTests(int channels) {
192 channels);
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| /external/webrtc/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/ |
| WebRtcAudioManager.java | 28 // fundamental audio parameters like native sample rate and number of channels.
62 private static final int CHANNELS = 1;
85 private int channels; field in class:WebRtcAudioManager
100 sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS,
139 channels = CHANNELS;
147 getMinOutputFrameSize(sampleRate, channels);
149 inputBufferSize = getMinInputFrameSize(sampleRate, channels);
269 assertTrue(numChannels == CHANNELS);
289 int sampleRate, int channels, boolean hardwareAEC, boolean hardwareAGC
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| /hardware/interfaces/audio/effect/2.0/ |
| types.hal | 222 * samples for all channels at a given time. Frame size for unspecified format
245 CHANNELS = 0x0004, // channels
248 ALL = BUFFER | SMP_RATE | CHANNELS | FORMAT | ACC_MODE
258 AudioChannelMask channels;
271 AUX_CHANNELS, // supports auxiliary channels
277 AudioChannelMask mainChannels; // channel mask for main channels
278 AudioChannelMask auxChannels; // channel mask for auxiliary channels
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| /hardware/interfaces/audio/effect/4.0/ |
| types.hal | 222 * samples for all channels at a given time. Frame size for unspecified format
245 CHANNELS = 0x0004, // channels
258 bitfield<AudioChannelMask> channels;
271 AUX_CHANNELS, // supports auxiliary channels
277 bitfield<AudioChannelMask> mainChannels; // channel mask for main channels
278 bitfield<AudioChannelMask> auxChannels; // channel mask for auxiliary channels
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| /external/walt/android/WALT/app/src/main/jni/ |
| player.c | 58 #define CHANNELS 1 // 1 for mono, 2 for stereo
257 // because when channels = 2 then there are 2 samples per frame.
406 format_pcm.numChannels = CHANNELS;
408 // because when channels = 2 then there are 2 samples per frame.
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| /cts/tests/camera/src/android/hardware/camera2/cts/ |
| AllocationTest.java | 283 * <p>The color channels must be in the following order:
309 final int CHANNELS = 3; // yuv
312 assertTrue("YUV pixel must be at least 3 bytes large", CHANNELS <= yuvData.length);
314 float[] rgb = new float[CHANNELS];
331 for (int i = 0; i < CHANNELS; ++i) {
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| ImageReaderTest.java | 585 final int CHANNELS = 3; // yuv
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| /hardware/interfaces/audio/common/2.0/ |
| types.hal | 352 /** output channels */
413 /** input channels */
464 * For multi-channel beyond stereo, the platform convention is that channels
761 CHANNELS = 0x2, // supports separate channel gain control
771 AudioChannelMask channelMask; // channels which gain an be controlled
787 AudioChannelMask channelMask; // channels which gain value follows
790 * 8 is not "FCC_8", so it won't need to be changed for > 8 channels.
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| /hardware/interfaces/audio/common/4.0/ |
| types.hal | 357 /** output channels */
430 /** input channels */
722 CHANNELS = 0x2, // supports separate channel gain control
732 bitfield<AudioChannelMask> channelMask; // channels which gain an be controlled
748 AudioChannelMask channelMask; // channels which gain value follows
751 * 8 is not "FCC_8", so it won't need to be changed for > 8 channels.
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