| /system/bt/embdrv/sbc/encoder/include/ |
| sbc_if.h | 29 bits_per_sample - 8 or 16
33 int SBC_init(int pcm_sample_freq, int channels, int bits_per_sample);
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| /hardware/intel/common/libmix/mix_audio/src/ |
| mixacp.c | 51 self->bits_per_sample = MIX_ACP_BPS_16;
145 t->bits_per_sample = s->bits_per_sample;
198 (acp1->bits_per_sample == acp2->bits_per_sample) &&
255 return obj->bits_per_sample;
273 obj->bits_per_sample = type;
313 mix_log(MIX_AUDIO_COMP, MIX_LOG_LEVEL_INFO, "bits_per_sample: %d\n", obj->bits_per_sample);
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| mixacp.h | 127 * @bits_per_sample: Number of output bit per sample. See #mix_acp_set_bps
148 MixACPBPSType bits_per_sample; member in struct:_MixAudioConfigParams
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| /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,
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| 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);
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| /external/tinyalsa/ |
| tinycap.c | 54 uint16_t bits_per_sample; member in struct:wav_header
157 header.bits_per_sample = pcm_format_to_bits(format);
158 header.byte_rate = (header.bits_per_sample / 8) * channels * rate;
159 header.block_align = channels * (header.bits_per_sample / 8);
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| tinyplay.c | 58 uint16_t bits_per_sample; member in struct:chunk_fmt
156 chunk_fmt.bits_per_sample, period_size, period_count);
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| 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_enc_dec.h | 68 int bits_per_sample; member in struct:__anon26891
106 int bits_per_sample; member in struct:__anon26893
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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;
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| /external/flac/libFLAC/include/protected/ |
| stream_decoder.h | 46 unsigned bits_per_sample; member in struct:FLAC__StreamDecoderProtected
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| stream_encoder.h | 94 unsigned bits_per_sample; member in struct:FLAC__StreamEncoderProtected
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| /external/webrtc/talk/media/base/ |
| audiorenderer.h | 42 int bits_per_sample,
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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
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| /system/extras/sound/ |
| playwav.c | 118 uint16_t bits_per_sample; member in struct:wav_header
168 hdr.num_channels, hdr.sample_rate, hdr.bits_per_sample,
182 if (hdr.bits_per_sample != 16) {
213 hdr.bits_per_sample = 16;
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| /external/flac/libFLAC/ |
| stream_encoder_framing.c | 91 FLAC__ASSERT(metadata->data.stream_info.bits_per_sample > 0);
92 FLAC__ASSERT(metadata->data.stream_info.bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
93 if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.bits_per_sample-1, FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN))
310 FLAC__ASSERT(header->bits_per_sample > 0 && header->bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
311 switch(header->bits_per_sample) {
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| /hardware/qcom/audio/legacy/libalsa-intf/ |
| arec.c | 82 uint16_t bits_per_sample; member in struct:wav_header
450 hdr.bits_per_sample = 16;
451 hdr.byte_rate = (rate * ch * hdr.bits_per_sample) / 8;
452 hdr.block_align = ( hdr.bits_per_sample * ch ) / 8;
470 hdr.num_channels, hdr.sample_rate, hdr.bits_per_sample,
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| alsa_audio.h | 205 __u32 bits_per_sample; member in struct:snd_enc_wma
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| /external/webrtc/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/ |
| WebRtcAudioManager.java | 57 private static final int BITS_PER_SAMPLE = 16;
244 final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
268 final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
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| WebRtcAudioRecord.java | 34 private static final int BITS_PER_SAMPLE = 16;
165 final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
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| WebRtcAudioTrack.java | 32 private static final int BITS_PER_SAMPLE = 16;
158 final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
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| /external/webrtc/talk/app/webrtc/ |
| rtpsender.cc | 45 int bits_per_sample,
51 sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
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| rtpsender.h | 58 int bits_per_sample,
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| /frameworks/base/cmds/bootanimation/ |
| AudioPlayer.cpp | 53 uint16_t bits_per_sample; member in struct:chunk_fmt
277 if (chunkFmt->bits_per_sample != 16) {
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| 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);
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