| /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...] |
| Android.mk | 11 resampler.c \
|
| 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
124 /* additional space in resampler buffer allowing for extra samples to be returned
125 * by speex resampler when sample rates ratio is not an integer.
163 if (er->resampler != NULL) {
164 er->resampler->reset(er->resampler);
[all...] |
| /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
|
| /external/webkit/Source/WebCore/platform/audio/ |
| AudioResamplerKernel.cpp | 40 AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
41 : m_resampler(resampler)
|
| AudioBus.cpp | 415 SincResampler resampler(sampleRateRatio);
416 resampler.process(source, destination, sourceLength);
|
| /frameworks/av/services/audioflinger/ |
| AudioMixer.h | 81 // This clears out the resampler's input buffer.
184 AudioResampler* resampler; member in struct:android::AudioMixer::track_t
200 bool doesResample() const { return resampler != NULL; }
201 void resetResampler() { if (resampler != NULL) resampler->reset(); }
203 size_t getUnreleasedFrames() const { return resampler != NULL ?
204 resampler->getUnreleasedFrames() : 0; };
|
| AudioResampler.cpp | 104 if (property_get("af.resampler.quality", value, NULL) > 0) {
142 // read the resampler default quality property the first time it is needed
153 // naive implementation of CPU load throttling doesn't account for whether resampler is active
159 ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d",
184 AudioResampler* resampler; local
190 ALOGV("Create linear Resampler");
191 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
195 ALOGV("Create cubic Resampler");
196 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
200 ALOGV("Create HIGH_QUALITY sinc Resampler");
[all...] |
| AudioMixer.cpp | 126 t->resampler = NULL;
158 delete t->resampler;
201 t->resampler = NULL;
371 // delete the resampler
372 delete track.resampler;
373 track.resampler = NULL;
473 delete track.resampler;
474 track.resampler = NULL;
538 if (value != devSampleRate || resampler != NULL) {
541 if (resampler == NULL)
[all...] |
| /device/asus/grouper/audio/ |
| audio_hw.c | 37 #include <audio_utils/resampler.h>
120 struct resampler_itfe *resampler; member in struct:stream_out
140 struct resampler_itfe *resampler; member in struct:stream_in
219 if (out->resampler) {
220 release_resampler(out->resampler);
221 out->resampler = NULL;
240 if (in->resampler) {
241 release_resampler(in->resampler);
242 in->resampler = NULL;
302 * create a resampler
[all...] |
| /external/webrtc/src/common_audio/resampler/ |
| Android.mk | 20 LOCAL_SRC_FILES := resampler.cc
|
| /device/ti/panda/audio/ |
| audio_hw.c | 45 #include <audio_utils/resampler.h>
606 struct resampler_itfe *resampler; member in struct:omap4_stream_out
625 struct resampler_itfe *resampler; member in struct:omap4_stream_in
[all...] |
| /external/webrtc/ |
| Android.mk | 12 include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/Android.mk
|
| /device/samsung/tuna/audio/ |
| audio_hw.c | 35 #include <audio_utils/resampler.h>
681 struct resampler_itfe *resampler; member in struct:tuna_stream_out
720 struct resampler_itfe *resampler; member in struct:tuna_stream_in
[all...] |
| /device/samsung/manta/audio/ |
| audio_hw.c | 41 #include <audio_utils/resampler.h>
172 struct resampler_itfe *resampler; member in struct:stream_in
671 /* if no supported sample rate is available, use the resampler */
672 if (in->resampler)
673 in->resampler->reset(in->resampler);
835 if (in->resampler != NULL) {
836 in->resampler->resample_from_provider(in->resampler,
856 * in->resampler->resample_from_provider() *
[all...] |
| /external/webrtc/src/modules/audio_processing/aec/ |
| echo_cancellation.c | 84 void *resampler; member in struct:__anon16653
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);
|
| /cts/suite/pts/hostTests/browser/browserlauncher/assets/octane/ |
| gbemu.js | 318 // Start of js/other/resampler.js file.
320 //JavaScript Audio Resampler (c) 2011 - Grant Galitz
321 function Resampler(fromSampleRate, toSampleRate, channels, outputBufferSize, noReturn) {
329 Resampler.prototype.initialize = function () {
333 //Setup a resampler bypass:
334 this.resampler = this.bypassResampler; //Resampler just returns what was passed through.
338 //Setup the interpolation resampler:
340 this.resampler = this.interpolate; //Resampler is a custom quality interpolation algorithm
[all...] |
