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  /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));
sinc_resampler_unittest.cc 21 #include "webrtc/common_audio/resampler/sinc_resampler.h"
22 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
60 SincResampler resampler(kSampleRateRatio, SincResampler::kDefaultRequestSize,
64 int max_chunk_size = resampler.ChunkSize() * kChunks;
70 resampler.Resample(resampler.ChunkSize(), resampled_destination.get());
76 resampler.Resample(max_chunk_size, resampled_destination.get());
82 SincResampler resampler(kSampleRateRatio, SincResampler::kDefaultRequestSize,
84 scoped_ptr<float[]> resampled_destination(new float[resampler.ChunkSize()]);
86 // Fill the resampler with junk data
    [all...]
push_sinc_resampler_unittest.cc 16 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
17 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
85 PushSincResampler resampler(input_samples, output_samples);
90 resampler.Resample(source_int.get(),
98 resampler.Resample(source.get(),
138 PushSincResampler resampler(input_block_size, output_block_size);
148 // The sinc resampler has an implicit delay of approximately half the kernel
155 resampler.get_resampler_for_testing()->ChunkSize();
166 resampler.Resample(source_int.get(),
178 resampler.Resample(&source[i * input_block_size]
    [all...]
  /external/chromium_org/media/base/
sinc_resampler_perftest.cc 31 SincResampler* resampler,
37 convolve_fn(resampler->get_kernel_for_testing() + (aligned ? 0 : 1),
38 resampler->get_kernel_for_testing(),
39 resampler->get_kernel_for_testing(),
55 SincResampler resampler(kSampleRateRatio,
60 &resampler, SincResampler::Convolve_C, true, "unoptimized_aligned");
64 &resampler, SincResampler::CONVOLVE_FUNC, true, "optimized_aligned");
66 &resampler, SincResampler::CONVOLVE_FUNC, false, "optimized_unaligned");
sinc_resampler_unittest.cc 49 SincResampler resampler(
54 int max_chunk_size = resampler.ChunkSize() * kChunks;
60 resampler.Resample(resampler.ChunkSize(), resampled_destination.get());
66 resampler.Resample(max_chunk_size, resampled_destination.get());
72 SincResampler resampler(
75 scoped_ptr<float[]> resampled_destination(new float[resampler.ChunkSize()]);
77 // Fill the resampler with junk data.
80 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 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...]
Android.mk 14 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
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  /external/chromium_org/third_party/webrtc/voice_engine/
utility.h 18 #include "webrtc/common_audio/resampler/include/push_resampler.h"
32 PushResampler<int16_t>* resampler,
51 PushResampler<int16_t>* resampler,
utility.cc 13 #include "webrtc/common_audio/resampler/include/push_resampler.h"
28 PushResampler<int16_t>* resampler,
43 if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
53 int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
81 PushResampler<int16_t>* resampler,
101 if (resampler->InitializeIfNeeded(
112 int out_length = resampler->Resample(
  /frameworks/av/services/audioflinger/audio-resampler/
Android.mk 8 LOCAL_MODULE := libaudio-resampler
  /external/chromium_org/third_party/webrtc/modules/audio_processing/aec/
echo_cancellation_internal.h 51 void* resampler; member in struct:__anon20505
  /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
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Android.mk 4 # resampler unit test
  /frameworks/av/services/audioflinger/
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...]
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");
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  /device/htc/flounder/audio/hal/
Android.mk 10 # TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8
  /external/chromium_org/third_party/WebKit/Source/platform/audio/
AudioResamplerKernel.cpp 38 AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
39 : m_resampler(resampler)
  /external/chromium_org/third_party/webrtc/common_audio/
common_audio.target.darwin-arm.mk 28 third_party/webrtc/common_audio/resampler/push_resampler.cc \
29 third_party/webrtc/common_audio/resampler/push_sinc_resampler.cc \
30 third_party/webrtc/common_audio/resampler/resampler.cc \
31 third_party/webrtc/common_audio/resampler/sinc_resampler.cc \
177 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
302 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
common_audio.target.darwin-arm64.mk 28 third_party/webrtc/common_audio/resampler/push_resampler.cc \
29 third_party/webrtc/common_audio/resampler/push_sinc_resampler.cc \
30 third_party/webrtc/common_audio/resampler/resampler.cc \
31 third_party/webrtc/common_audio/resampler/sinc_resampler.cc \
163 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
273 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
common_audio.target.darwin-mips.mk 28 third_party/webrtc/common_audio/resampler/push_resampler.cc \
29 third_party/webrtc/common_audio/resampler/push_sinc_resampler.cc \
30 third_party/webrtc/common_audio/resampler/resampler.cc \
31 third_party/webrtc/common_audio/resampler/sinc_resampler.cc \
166 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
280 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
common_audio.target.darwin-mips64.mk 28 third_party/webrtc/common_audio/resampler/push_resampler.cc \
29 third_party/webrtc/common_audio/resampler/push_sinc_resampler.cc \
30 third_party/webrtc/common_audio/resampler/resampler.cc \
31 third_party/webrtc/common_audio/resampler/sinc_resampler.cc \
166 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
280 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
common_audio.target.darwin-x86.mk 28 third_party/webrtc/common_audio/resampler/push_resampler.cc \
29 third_party/webrtc/common_audio/resampler/push_sinc_resampler.cc \
30 third_party/webrtc/common_audio/resampler/resampler.cc \
31 third_party/webrtc/common_audio/resampler/sinc_resampler.cc \
169 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
285 $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \

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