| /external/chromium_org/media/base/ |
| sinc_resampler_unittest.cc | 57 SincResampler resampler(
62 int max_chunk_size = resampler.ChunkSize() * kChunks;
68 resampler.Resample(resampler.ChunkSize(), resampled_destination.get());
74 resampler.Resample(max_chunk_size, resampled_destination.get());
80 SincResampler resampler(
83 scoped_ptr<float[]> resampled_destination(new float[resampler.ChunkSize()]);
85 // Fill the resampler with junk data.
88 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get())
[all...] |
| multi_channel_resampler_unittest.cc | 31 // Chosen arbitrarily based on what each resampler reported during testing.
66 MultiChannelResampler resampler(
70 // First prime the resampler with some junk data, so we can verify Flush().
72 resampler.Resample(1, audio_bus_.get());
73 resampler.Flush();
81 resampler.Resample(frames, audio_bus_.get());
|
| /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 | 25 #include <audio_utils/resampler.h>
54 void *wr_src_buf; // resampler input buf (either wr_buf or buffer used by write())
63 struct resampler_itfe *resampler; // input resampler member in struct:echo_reference
64 struct resampler_buffer_provider provider; // resampler buffer provider
126 /* additional space in resampler buffer allowing for extra samples to be returned
127 * by speex resampler when sample rates ratio is not an integer.
165 if (er->resampler != NULL) {
166 er->resampler->reset(er->resampler);
[all...] |
| Android.mk | 11 resampler.c \
|
| /system/media/audio_utils/include/audio_utils/ |
| resampler.h | 41 /* call back interface used by the resampler to get new data */
61 /* resampler interface */
64 * reset resampler state
66 void (*reset)(struct resampler_itfe *resampler);
71 int (*resample_from_provider)(struct resampler_itfe *resampler,
79 int (*resample_from_input)(struct resampler_itfe *resampler,
85 * return the latency introduced by the resampler in ns.
87 int32_t (*delay_ns)(struct resampler_itfe *resampler);
91 * create a resampler according to input parameters passed.
103 * release resampler resources
[all...] |
| /frameworks/av/services/audioflinger/audio-resampler/ |
| Android.mk | 8 LOCAL_MODULE := libaudio-resampler
|
| /frameworks/av/services/audioflinger/ |
| test-resample.cpp | 70 fprintf(stderr," -q resampler quality\n");
213 AudioResampler* resampler = AudioResampler::create(16, channels, local
217 resampler->setSampleRate(input_freq);
218 resampler->setVolume(0x1000, 0x1000);
223 resampler->resample((int*) output_vaddr, out_frames, &provider);
224 resampler->resample((int*) output_vaddr, out_frames, &provider);
225 resampler->resample((int*) output_vaddr, out_frames, &provider);
226 resampler->resample((int*) output_vaddr, out_frames, &provider);
233 delete resampler;
236 AudioResampler* resampler = AudioResampler::create(16, channels local
[all...] |
| AudioMixer.h | 89 // This clears out the resampler's input buffer.
193 AudioResampler* resampler; member in struct:android::AudioMixer::track_t
209 bool doesResample() const { return resampler != NULL; }
210 void resetResampler() { if (resampler != NULL) resampler->reset(); }
212 size_t getUnreleasedFrames() const { return resampler != NULL ?
213 resampler->getUnreleasedFrames() : 0; };
|
| AudioResampler.cpp | 102 if (property_get("af.resampler.quality", value, NULL) > 0) {
140 // read the resampler default quality property the first time it is needed
151 // naive implementation of CPU load throttling doesn't account for whether resampler is active
157 ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
182 AudioResampler* resampler; local
188 ALOGV("Create linear Resampler");
189 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
192 ALOGV("Create cubic Resampler");
193 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
196 ALOGV("Create HIGH_QUALITY sinc Resampler");
[all...] |
| Android.mk | 72 # build audio resampler test tool
|
| AudioMixer.cpp | 134 t->resampler = NULL;
166 delete t->resampler;
213 t->resampler = NULL;
383 // delete the resampler
384 delete track.resampler;
385 track.resampler = NULL;
485 delete track.resampler;
486 track.resampler = NULL;
550 if (value != devSampleRate || resampler != NULL) {
553 if (resampler == NULL)
[all...] |
| /external/chromium_org/third_party/WebKit/Source/core/platform/audio/ |
| AudioResamplerKernel.cpp | 40 AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
41 : m_resampler(resampler)
|
| AudioBus.cpp | 580 SincResampler resampler(sampleRateRatio);
581 resampler.process(source, destination, sourceLength);
|
| /device/asus/grouper/audio/ |
| audio_hw.c | 37 #include <audio_utils/resampler.h>
122 struct resampler_itfe *resampler; member in struct:stream_out
142 struct resampler_itfe *resampler; member in struct:stream_in
221 if (out->resampler) {
222 release_resampler(out->resampler);
223 out->resampler = NULL;
242 if (in->resampler) {
243 release_resampler(in->resampler);
244 in->resampler = NULL;
304 * create a resampler
[all...] |
| /external/webrtc/src/common_audio/resampler/ |
| Android.mk | 20 LOCAL_SRC_FILES := resampler.cc
|
| /device/samsung/manta/audio/ |
| audio_hw.c | 41 #include <audio_utils/resampler.h>
177 struct resampler_itfe *resampler; member in struct:stream_in
759 /* if no supported sample rate is available, use the resampler */
760 if (in->resampler)
761 in->resampler->reset(in->resampler);
874 if (in->resampler != NULL) {
875 in->resampler->resample_from_provider(in->resampler,
895 * in->resampler->resample_from_provider() *
[all...] |
| /external/chromium_org/third_party/opus/src/ |
| silk_sources.mk | 63 silk/resampler.c \
|
| /external/webrtc/src/modules/audio_processing/aec/ |
| echo_cancellation.c | 84 void *resampler; member in struct:__anon30471
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/flo/ |
| device-common.mk | 127 af.resampler.quality=4
172 libaudio-resampler
|
| /device/lge/mako/ |
| device.mk | 142 af.resampler.quality=4
182 libaudio-resampler
|
| /external/webrtc/ |
| Android.mk | 12 include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/Android.mk
|
| /device/lge/hammerhead/ |
| device.mk | 143 libaudio-resampler
300 af.resampler.quality=4
|
| /external/chromium_org/third_party/opus/ |
| opus.target.darwin-arm.mk | 114 third_party/opus/src/silk/resampler.c \
|
| opus.target.darwin-mips.mk | 114 third_party/opus/src/silk/resampler.c \
|
