| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_manager.h | 102 jint sample_rate,
112 jint sample_rate,
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| audio_manager_unittest.cc | 84 PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate());
91 PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate());
121 EXPECT_EQ(0, params.sample_rate());
141 EXPECT_EQ(kSampleRate, params.sample_rate());
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| audio_manager.cc | 177 jint sample_rate,
189 env, sample_rate, channels, hardware_aec, hardware_agc, hardware_ns,
194 jint sample_rate,
207 ALOGD("sample_rate: %d", sample_rate);
217 playout_parameters_.reset(sample_rate, static_cast<size_t>(channels),
219 record_parameters_.reset(sample_rate, static_cast<size_t>(channels),
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| /frameworks/av/services/audioflinger/ |
| AudioHwDevice.cpp | 50 config->sample_rate,
65 config->sample_rate,
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| /hardware/qcom/msm8994/kernel-headers/linux/ |
| msm_audio_aac.h | 71 unsigned short sample_rate; member in struct:msm_audio_aac_config
76 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
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| /hardware/qcom/msm8994/original-kernel-headers/linux/ |
| msm_audio_aac.h | 66 unsigned short sample_rate; member in struct:msm_audio_aac_config
71 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
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| /hardware/qcom/msm8996/kernel-headers/linux/ |
| msm_audio_aac.h | 71 unsigned short sample_rate; member in struct:msm_audio_aac_config
76 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
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| /hardware/qcom/msm8996/original-kernel-headers/linux/ |
| msm_audio_aac.h | 66 unsigned short sample_rate; member in struct:msm_audio_aac_config
71 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
|
| /hardware/qcom/msm8998/kernel-headers/linux/ |
| msm_audio_aac.h | 71 unsigned short sample_rate; member in struct:msm_audio_aac_config
76 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
|
| /hardware/qcom/msm8998/original-kernel-headers/linux/ |
| msm_audio_aac.h | 66 unsigned short sample_rate; member in struct:msm_audio_aac_config
71 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
|
| /hardware/qcom/msm8x84/kernel-headers/linux/ |
| msm_audio_aac.h | 71 unsigned short sample_rate; member in struct:msm_audio_aac_config
76 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
|
| /hardware/qcom/msm8x84/original-kernel-headers/linux/ |
| msm_audio_aac.h | 66 unsigned short sample_rate; member in struct:msm_audio_aac_config
71 uint32_t sample_rate; member in struct:msm_audio_aac_enc_config
|
| /packages/apps/TV/proto/ |
| track.proto | 30 optional int32 sample_rate = 5;
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| /system/media/alsa_utils/include/ |
| alsa_device_profile.h | 89 unsigned profile_calc_min_period_size(alsa_device_profile* profile, unsigned sample_rate);
90 unsigned int profile_get_period_size(alsa_device_profile* profile, unsigned sample_rate);
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| /external/webrtc/webrtc/common_audio/ |
| wav_file.cc | 43 s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
45 << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
101 WavWriter::WavWriter(const std::string& filename, int sample_rate,
103 : sample_rate_(sample_rate),
155 int sample_rate,
158 new webrtc::WavWriter(filename, sample_rate, num_channels));
172 return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
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| /system/extras/sound/ |
| playwav.c | 28 uint32_t sample_rate; member in struct:msm_audio_config
60 config.sample_rate = rate;
115 uint32_t sample_rate; member in struct:wav_header
116 uint32_t byte_rate; /* sample_rate * num_channels * bps / 8 */
168 hdr.num_channels, hdr.sample_rate, hdr.bits_per_sample,
187 play_file(hdr.sample_rate, hdr.num_channels,
210 hdr.sample_rate = rate;
211 hdr.byte_rate = hdr.sample_rate * hdr.num_channels * 2;
238 cfg.sample_rate = hdr.sample_rate;
[all...] |
| /device/google/dragon/audio/hal/ |
| cras_dsp_pipeline.h | 59 * sample_rate - The audio sampling rate.
63 int cras_dsp_pipeline_instantiate(struct pipeline *pipeline, int sample_rate);
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| /external/webrtc/webrtc/modules/audio_coding/test/ |
| PacketLossTest.h | 41 std::string in_file_name, int sample_rate, int channels,
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| /external/webrtc/webrtc/voice_engine/ |
| voe_base_impl.cc | 114 const int16_t* audio_data, int sample_rate,
123 voe_channels, number_of_voe_channels, audio_data, sample_rate,
134 PushCaptureData(voe_channels[i], audio_data, 16, sample_rate,
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,
159 sample_rate, number_of_frames, number_of_channels); local
160 channel_ptr->PrepareEncodeAndSend(sample_rate);
166 int sample_rate,
172 assert(number_of_frames == static_cast<size_t>(sample_rate / 100))
[all...] |
| /hardware/libhardware/modules/vehicle/ |
| vehicle.c | 56 float sample_rate; member in struct:subscription
315 ALOGD("prop: %d rate: %f", sub->prop, sub->sample_rate);
408 static int vdev_subscribe(vehicle_hw_device_t* device, int32_t prop, float sample_rate,
410 ALOGD("vdev_subscribe 0x%x, %f", prop, sample_rate);
436 if ((config->change_mode == VEHICLE_PROP_CHANGE_MODE_ON_CHANGE) && (sample_rate != 0)) {
439 prop, sample_rate);
442 if ((config->max_sample_rate < sample_rate) || (config->min_sample_rate > sample_rate)) {
445 prop, sample_rate, config->min_sample_rate, config->max_sample_rate);
453 sub->sample_rate = sample_rate
[all...] |
| /external/flac/libFLAC/ |
| format.c | 200 FLAC_API FLAC__bool FLAC__format_sample_rate_is_valid(unsigned sample_rate)
202 if(sample_rate == 0 || sample_rate > FLAC__MAX_SAMPLE_RATE) {
209 FLAC_API FLAC__bool FLAC__format_blocksize_is_subset(unsigned blocksize, unsigned sample_rate)
213 else if(sample_rate <= 48000 && blocksize > 4608)
219 FLAC_API FLAC__bool FLAC__format_sample_rate_is_subset(unsigned sample_rate)
222 !FLAC__format_sample_rate_is_valid(sample_rate) ||
224 sample_rate >= (1u << 16) &&
225 !(sample_rate % 1000 == 0 || sample_rate % 10 == 0
[all...] |
| /system/bt/audio_a2dp_hw/src/ |
| audio_a2dp_hw.cc | 464 tA2DP_SAMPLE_RATE sample_rate; local
472 if (a2dp_ctrl_receive(common, &sample_rate, sizeof(tA2DP_SAMPLE_RATE)) < 0)
479 switch (sample_rate) {
482 common->cfg.rate = sample_rate;
485 ERROR("Invalid sample rate: %" PRIu32, sample_rate);
520 if (a2dp_ctrl_receive(common, &codec_config->sample_rate,
534 if (a2dp_ctrl_receive(common, &codec_capability->sample_rate,
548 switch (codec_config->sample_rate) {
569 ERROR("Invalid sample rate: 0x%x", codec_config->sample_rate);
610 codec_config->sample_rate, codec_config->bits_per_sample
935 uint32_t sample_rate; local
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| neteq_stereo_unittest.cc | 29 int sample_rate; member in struct:webrtc::TestParameters
51 sample_rate_hz_(GetParam().sample_rate),
383 int sample_rate = sample_rates[rate_index]; local
388 p.sample_rate = sample_rate;
391 if (sample_rate == 8000) {
405 ", sample_rate = " << p.sample_rate << "}";
|
| /system/bt/stack/a2dp/ |
| a2dp_sbc.cc | [all...] |
| /device/generic/goldfish/audio/ |
| audio_hw_legacy.c | 57 uint32_t sample_rate; member in struct:generic_stream_out
70 return out->sample_rate;
401 if (config->sample_rate < sample_rates[idx]) {
402 config->sample_rate = sample_rates[idx];
403 ALOGE("Error opening output stream, sample_rate %u", config->sample_rate);
406 } else if (config->sample_rate == sample_rates[idx]) {
410 config->sample_rate = sample_rates[idx];
411 ALOGE("Error opening output stream, sample_rate %u", config->sample_rate);
[all...] |
