| /external/webrtc/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/webrtc/webrtc/modules/audio_processing/aec/ |
| echo_cancellation_internal.h | 51 void* resampler; member in struct:__anon27006
|
| /external/webrtc/webrtc/voice_engine/ |
| utility_unittest.cc | 15 #include "webrtc/common_audio/resampler/include/push_resampler.h"
130 PushResampler<int16_t> resampler; // Create a new one with every test. local
157 // The sinc resampler has a known delay, which we compute here. Multiplying by
158 // two gives us a crude maximum for any resampling, as the old resampler
166 RemixAndResample(src_frame_, &resampler, &dst_frame_);
169 // The sinc resampler gives poor SNR at this extreme conversion, but we
|
| /frameworks/av/services/audioflinger/ |
| AudioResampler.cpp | 108 if (property_get("af.resampler.quality", value, NULL) > 0) {
152 // read the resampler default quality property the first time it is needed
163 /* if the caller requests DEFAULT_QUALITY and af.resampler.property
164 * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
173 // naive implementation of CPU load throttling doesn't account for whether resampler is active
179 ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
212 AudioResampler* resampler; local
217 ALOGV("Create linear Resampler");
219 resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);
222 ALOGV("Create cubic Resampler");
[all...] |
| test-resample.cpp | 49 fprintf(stderr," -q resampler quality\n");
343 AudioResampler* resampler = AudioResampler::create(format, channels, local
349 resampler->setSampleRate(9000);
350 resampler->setSampleRate(12000);
351 resampler->setSampleRate(20000);
352 resampler->setSampleRate(30000);
364 resampler->setSampleRate(1000);
368 resampler->setSampleRate(1000+i);
376 resampler->reset();
377 delete resampler;
381 AudioResampler* resampler = AudioResampler::create(format, channels, local
[all...] |
| AudioMixer.h | 97 // This clears out the resampler's input buffer.
213 AudioResampler* resampler; member in struct:android::AudioMixer::track_t
267 bool doesResample() const { return resampler != NULL; }
268 void resetResampler() { if (resampler != NULL) resampler->reset(); }
270 size_t getUnreleasedFrames() const { return resampler != NULL ?
271 resampler->getUnreleasedFrames() : 0; };
|
| /frameworks/av/services/audioflinger/tests/ |
| resampler_tests.cpp | 41 android::AudioBufferProvider *provider, android::AudioResampler *resampler)
51 size_t framesResampled = resampler->resample(
95 // create the resampler
96 android::AudioResampler* resampler; local
98 resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
99 resampler->setSampleRate(inputFreq);
100 resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
107 resample(channels, reference, outputFrames, refIncr, &provider, resampler);
113 resampler->reset();
115 delete resampler;
182 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...] |
| /device/google/dragon/audio/hal/ |
| audio_hw.h | 24 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
25 #include <audio_utils/resampler.h>
209 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
210 struct resampler_itfe* resampler; member in struct:pcm_device
263 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
265 struct resampler_itfe* resampler; member in struct:stream_in
|
| /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>
259 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
260 struct resampler_itfe* resampler; member in struct:pcm_device
331 /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
333 struct resampler_itfe* resampler; member in struct:stream_in
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| audio_processing_unittest.cc | 20 #include "webrtc/common_audio/resampler/include/push_resampler.h"
21 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
2574 PushResampler<float> resampler; local
[all...] |
