| /external/chromium_org/third_party/webrtc/common_audio/resampler/ |
| push_resampler_unittest.cc | 12 #include "webrtc/common_audio/resampler/include/push_resampler.h"
19 PushResampler<int16_t> resampler; local
20 EXPECT_EQ(-1, resampler.InitializeIfNeeded(-1, 16000, 1));
21 EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, -1, 1));
22 EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 0));
23 EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 3));
24 EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1));
25 EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2));
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/aec/ |
| echo_cancellation_internal.h | 51 void* resampler; member in struct:__anon20505
|
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| utility_unittest.cc | 14 #include "webrtc/common_audio/resampler/include/push_resampler.h"
134 PushResampler<int16_t> resampler; // Create a new one with every test. local
161 // The sinc resampler has a known delay, which we compute here. Multiplying by
162 // two gives us a crude maximum for any resampling, as the old resampler
170 RemixAndResample(src_frame_, &resampler, &dst_frame_);
180 &resampler,
185 // The sinc resampler gives poor SNR at this extreme conversion, but we
|
| /frameworks/av/services/audioflinger/ |
| AudioResampler.cpp | 110 if (property_get("af.resampler.quality", value, NULL) > 0) {
154 // read the resampler default quality property the first time it is needed
165 /* if the caller requests DEFAULT_QUALITY and af.resampler.property
166 * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
175 // naive implementation of CPU load throttling doesn't account for whether resampler is active
181 ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
214 AudioResampler* resampler; local
219 ALOGV("Create linear Resampler");
221 resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);
224 ALOGV("Create cubic Resampler");
[all...] |
| test-resample.cpp | 49 fprintf(stderr," -q resampler quality\n");
345 AudioResampler* resampler = AudioResampler::create(format, channels, local
351 resampler->setSampleRate(9000);
352 resampler->setSampleRate(12000);
353 resampler->setSampleRate(20000);
354 resampler->setSampleRate(30000);
366 resampler->setSampleRate(1000);
370 resampler->setSampleRate(1000+i);
378 resampler->reset();
379 delete resampler;
383 AudioResampler* resampler = AudioResampler::create(format, channels, local
[all...] |
| AudioMixer.h | 94 // This clears out the resampler's input buffer.
202 AudioResampler* resampler; member in struct:android::AudioMixer::track_t
234 bool doesResample() const { return resampler != NULL; }
235 void resetResampler() { if (resampler != NULL) resampler->reset(); }
237 size_t getUnreleasedFrames() const { return resampler != NULL ?
238 resampler->getUnreleasedFrames() : 0; };
|
| /frameworks/av/services/audioflinger/tests/ |
| resampler_tests.cpp | 41 android::AudioBufferProvider *provider, android::AudioResampler *resampler)
51 resampler->resample((int32_t*) output + channels*i, thisFrames, provider);
92 // create the resampler
93 android::AudioResampler* resampler; local
95 resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
96 resampler->setSampleRate(inputFreq);
97 resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
104 resample(channels, reference, outputFrames, refIncr, &provider, resampler);
110 resampler->reset();
112 delete resampler;
179 android::AudioResampler* resampler; local
[all...] |
| /system/media/audio_utils/ |
| resampler.c | 18 #define LOG_TAG "resampler"
24 #include <audio_utils/resampler.h>
28 struct resampler { struct
30 SpeexResamplerState *speex_resampler; // handle on speex resampler
41 int32_t speex_delay_ns; // delay introduced by speex resampler in ns
46 // speex based resampler
49 static void resampler_reset(struct resampler_itfe *resampler)
51 struct resampler *rsmp = (struct resampler *)resampler;
[all...] |
| echo_reference.c | 27 #include <audio_utils/resampler.h>
56 void *wr_src_buf; // resampler input buf (either wr_buf or buffer used by write())
65 struct resampler_itfe *resampler; // input resampler member in struct:echo_reference
66 struct resampler_buffer_provider provider; // resampler buffer provider
128 /* additional space in resampler buffer allowing for extra samples to be returned
129 * by speex resampler when sample rates ratio is not an integer.
167 if (er->resampler != NULL) {
168 er->resampler->reset(er->resampler);
[all...] |
| /external/webrtc/src/modules/audio_processing/aec/ |
| echo_cancellation.c | 84 void *resampler; member in struct:__anon17141
121 if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
193 WebRtcAec_FreeResampler(aecpc->resampler);
226 if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
329 newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
434 retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
|
| /device/asus/grouper/audio/ |
| audio_hw.c | 37 #include <audio_utils/resampler.h>
133 struct resampler_itfe *resampler; member in struct:stream_out
154 struct resampler_itfe *resampler; member in struct:stream_in
233 if (out->resampler) {
234 release_resampler(out->resampler);
235 out->resampler = NULL;
254 if (in->resampler) {
255 release_resampler(in->resampler);
256 in->resampler = NULL;
316 * create a resampler
[all...] |
| /device/htc/flounder/audio/hal/ |
| audio_hw.h | 25 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
26 #include <audio_utils/resampler.h>
257 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
258 struct resampler_itfe* resampler; member in struct:pcm_device
323 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
325 struct resampler_itfe* resampler; member in struct:stream_in
|
| /device/samsung/manta/audio/ |
| audio_hw.c | 41 #include <audio_utils/resampler.h>
186 struct resampler_itfe *resampler; member in struct:stream_in
780 /* if no supported sample rate is available, use the resampler */
781 if (in->resampler)
782 in->resampler->reset(in->resampler);
899 if (in->resampler != NULL) {
900 in->resampler->resample_from_provider(in->resampler,
920 * in->resampler->resample_from_provider() *
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
| audio_processing_unittest.cc | 18 #include "webrtc/common_audio/resampler/include/push_resampler.h"
19 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
2259 PushResampler<float> resampler; local
[all...] |
