/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_resampler.cc | 11 #include "webrtc/common_audio/resampler/include/push_resampler.h" 16 #include "webrtc/common_audio/resampler/include/resampler.h" 17 #include "webrtc/common_audio/resampler/push_sinc_resampler.h" 75 // The old resampler provides this memcpy facility in the case of matching 76 // sample rates, so reproduce it here for the sinc resampler.
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sinusoidal_linear_chirp_source.h | 18 #include "webrtc/common_audio/resampler/sinc_resampler.h" 22 // Fake audio source for testing the resampler. Generates a sinusoidal linear 24 // resampler for the specific sample rate conversion being used.
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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...] |
/frameworks/av/services/audioflinger/audio-resampler/ |
Android.mk | 8 LOCAL_MODULE := libaudio-resampler
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/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/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/common_audio/ |
common_audio.gyp | 21 'resampler/include', 26 'resampler/include', 38 'resampler/include/push_resampler.h', 39 'resampler/include/resampler.h', 40 'resampler/push_resampler.cc', 41 'resampler/push_sinc_resampler.cc', 42 'resampler/push_sinc_resampler.h', 43 'resampler/resampler.cc' [all...] |
BUILD.gn | 14 "resampler/include", 28 "resampler/include/push_resampler.h", 29 "resampler/include/resampler.h", 30 "resampler/push_resampler.cc", 31 "resampler/push_sinc_resampler.cc", 32 "resampler/push_sinc_resampler.h", 33 "resampler/resampler.cc", 34 "resampler/sinc_resampler.cc" [all...] |
/external/webrtc/src/common_audio/ |
common_audio.gyp | 13 'resampler/resampler.gypi',
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/external/chromium_org/third_party/speex/include/speex/ |
speex_resampler.h | 46 /* If the resampler is defined outside of Speex, we change the symbol names so that 114 /** Create a new resampler with integer input and output rates. 120 * @return Newly created resampler state 129 /** Create a new resampler with fractional input/output rates. The sampling 139 * @return Newly created resampler state 150 /** Destroy a resampler state. 151 * @param st Resampler state 156 * @param st Resampler state 173 * @param st Resampler state 190 * @param st Resampler stat [all...] |
/external/speex/include/speex/ |
speex_resampler.h | 46 /* If the resampler is defined outside of Speex, we change the symbol names so that 114 /** Create a new resampler with integer input and output rates. 120 * @return Newly created resampler state 129 /** Create a new resampler with fractional input/output rates. The sampling 139 * @return Newly created resampler state 150 /** Destroy a resampler state. 151 * @param st Resampler state 156 * @param st Resampler state 173 * @param st Resampler state 190 * @param st Resampler stat [all...] |
/external/webrtc/src/common_audio/resampler/ |
resampler.gypi | 12 'target_name': 'resampler', 26 'include/resampler.h', 27 'resampler.cc', 38 'resampler',
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/external/chromium_org/third_party/webrtc/common_audio/resampler/include/ |
resampler.h | 64 class Resampler 68 Resampler(); 70 Resampler(int inFreq, int outFreq, ResamplerType type); 71 ~Resampler(); 110 Resampler* slave_left_; 111 Resampler* slave_right_;
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/external/webrtc/src/common_audio/resampler/include/ |
resampler.h | 64 class Resampler 68 Resampler(); 70 Resampler(int inFreq, int outFreq, ResamplerType type); 71 ~Resampler(); 110 Resampler* slave_left_; 111 Resampler* slave_right_;
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/docs/source.android.com/src/devices/ |
audio_src.jd | 39 another sample rate. A sample rate converter, or resampler, is a module 40 that implements sample rate conversion. With respect to the resampler, 49 internally. In that case, a resampler would be used to upsample the MP3 55 The characteristics of a resampler can be expressed using metrics, including: 62 <li>overall latency through the resampler</li> 71 The ideal resampler would exactly preserve the source signal's amplitude 84 Section <a href="#srcResamplers">Resampler implementations</a> 89 <h2 id="srcResamplers">Resampler implementations</h2> 92 Available resampler implementations change frequently, 113 The specific resampler implementation selected depends o [all...] |
/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...] |
/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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/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/webrtc/common_audio/signal_processing/ |
resample_48khz.c | 26 // 48 -> 16 resampler 52 // initialize state of 48 -> 16 resampler 64 // 16 -> 48 resampler 90 // initialize state of 16 -> 48 resampler 102 // 48 -> 8 resampler 134 // initialize state of 48 -> 8 resampler 147 // 8 -> 48 resampler 179 // initialize state of 8 -> 48 resampler
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
acm_opus.h | 14 #include "webrtc/common_audio/resampler/include/resampler.h"
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acm_resampler.cc | 16 #include "webrtc/common_audio/resampler/include/resampler.h"
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