/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));
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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");
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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());
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/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 \
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/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...] |
/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,
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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(
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/frameworks/av/services/audioflinger/audio-resampler/ |
Android.mk | 8 LOCAL_MODULE := libaudio-resampler
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/external/chromium_org/third_party/webrtc/modules/audio_processing/aec/ |
echo_cancellation_internal.h | 51 void* resampler; member in struct:__anon20505
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/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...] |
Android.mk | 4 # resampler unit test
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/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"); [all...] |
/device/htc/flounder/audio/hal/ |
Android.mk | 10 # TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8
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/external/chromium_org/third_party/WebKit/Source/platform/audio/ |
AudioResamplerKernel.cpp | 38 AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler) 39 : m_resampler(resampler)
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/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 \
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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 \
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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 \
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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 \
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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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