| /external/webrtc/talk/media/base/ |
| audiorenderer.h | 42 int bits_per_sample,
|
| /external/flac/libFLAC/include/protected/ |
| stream_decoder.h | 46 unsigned bits_per_sample; member in struct:FLAC__StreamDecoderProtected
|
| stream_encoder.h | 94 unsigned bits_per_sample; member in struct:FLAC__StreamEncoderProtected
|
| /external/u-boot/drivers/sound/ |
| wm8994.h | 60 * @param bits_per_sample bits per Sample can be 16 or 24
67 int bits_per_sample, unsigned int channels);
|
| max98095.c | 138 * @param bits_per_sample Bits per sample
144 unsigned int rate, unsigned int bits_per_sample)
162 switch (bits_per_sample) {
173 __func__, bits_per_sample);
477 int bits_per_sample)
508 bits_per_sample);
576 int bits_per_sample)
589 bits_per_sample);
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| wm8994.c | 277 * @param bits_per_sample Bits per sample
283 unsigned int sampling_rate, unsigned int bits_per_sample,
317 switch (bits_per_sample) {
869 int bits_per_sample, unsigned int channels)
906 bits_per_sample, channels);
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| /external/webrtc/webrtc/modules/audio_device/include/ |
| audio_device_defines.h | 104 int bits_per_sample,
117 int bits_per_sample,
126 virtual void PullRenderData(int bits_per_sample,
163 size_t bits_per_sample() const { return kBitsPerSample; } function in class:webrtc::AudioParameters
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| g722_enc_dec.h | 68 int bits_per_sample; member in struct:__anon49740
106 int bits_per_sample; member in struct:__anon49742
|
| g722_decode.c | 163 s->bits_per_sample = 6;
165 s->bits_per_sample = 7;
167 s->bits_per_sample = 8;
170 if ((options & G722_PACKED) && s->bits_per_sample != 8)
268 if (s->in_bits < s->bits_per_sample)
273 code = s->in_buffer & ((1 << s->bits_per_sample) - 1);
274 s->in_buffer >>= s->bits_per_sample;
275 s->in_bits -= s->bits_per_sample;
282 switch (s->bits_per_sample)
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| g722_encode.c | 164 s->bits_per_sample = 6;
166 s->bits_per_sample = 7;
168 s->bits_per_sample = 8;
171 if ((options & G722_PACKED) && s->bits_per_sample != 8)
378 code = (0xC0 | ilow) >> (8 - s->bits_per_sample);
411 code = ((ihigh << 6) | ilow) >> (8 - s->bits_per_sample);
418 s->out_bits += s->bits_per_sample;
|
| /external/webrtc/webrtc/voice_engine/ |
| voe_base_impl.h | 87 int bits_per_sample,
93 int bits_per_sample,
97 void PullRenderData(int bits_per_sample,
|
| voe_base_impl.cc | 143 int bits_per_sample, int sample_rate,
145 PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate,
150 int bits_per_sample, int sample_rate,
165 void VoEBaseImpl::PullRenderData(int bits_per_sample,
171 assert(bits_per_sample == 16);
|
| /external/tensorflow/tensorflow/core/lib/wav/ |
| wav_io.cc | 49 char bits_per_sample[2]; member in struct:tensorflow::wav::__anon45632::FormatChunk
194 core::EncodeFixed16(format_chunk->bits_per_sample, kBitsPerSample);
249 uint16 bits_per_sample; local
250 TF_RETURN_IF_ERROR(ReadValue<uint16>(wav_string, &bits_per_sample, &offset));
251 if (bits_per_sample != 16) {
253 "Can only read 16-bit WAV files, but received ", bits_per_sample);
256 ((bits_per_sample * *channel_count) + 7) / 8;
|
| /external/tinyalsa/ |
| tinycap.c | 55 uint16_t bits_per_sample; member in struct:wav_header
165 header.bits_per_sample = pcm_format_to_bits(format);
166 header.byte_rate = (header.bits_per_sample / 8) * channels * rate;
167 header.block_align = channels * (header.bits_per_sample / 8);
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| tinyplay.c | 59 uint16_t bits_per_sample; member in struct:chunk_fmt
158 chunk_fmt.bits_per_sample, period_size, period_count, chunk_header.sz);
|
| /external/webrtc/webrtc/modules/audio_device/android/ |
| opensles_player.h | 99 size_t bits_per_sample);
|
| opensles_player.cc | 57 audio_parameters_.bits_per_sample());
192 size_t bits_per_sample) {
194 RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
|
| /frameworks/av/media/libstagefright/flac/dec/ |
| FLACDecoder.h | 63 return mStreamInfo.bits_per_sample;
|
| /external/flac/libFLAC/ |
| stream_encoder_framing.c | 95 FLAC__ASSERT(metadata->data.stream_info.bits_per_sample > 0);
96 FLAC__ASSERT(metadata->data.stream_info.bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
97 if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.bits_per_sample-1, FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN))
315 FLAC__ASSERT(header->bits_per_sample > 0 && header->bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
316 switch(header->bits_per_sample) {
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| /external/webrtc/talk/app/webrtc/ |
| mediastreaminterface.h | 152 int bits_per_sample,
|
| rtpsender.h | 58 int bits_per_sample,
|
| rtpsender.cc | 45 int bits_per_sample,
51 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
|
| /external/ImageMagick/coders/ |
| tiff.c | 1284 bits_per_sample, local
3461 bits_per_sample, local
[all...] |
| /external/flac/include/FLAC/ |
| format.h | 424 unsigned bits_per_sample; member in struct:__anon22388
534 unsigned bits_per_sample; member in struct:__anon22393
[all...] |
| /device/google/bonito/sdm710/kernel-headers/sound/ |
| compress_params.h | 170 __u32 bits_per_sample; member in struct:snd_enc_wma
233 __u16 bits_per_sample; member in struct:snd_dec_ape
|
