| /external/webrtc/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 | 22 #include "webrtc/common_audio/resampler/sinc_resampler.h"
23 #include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
60 SincResampler resampler(kSampleRateRatio, SincResampler::kDefaultRequestSize,
64 size_t 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,
85 new float[resampler.ChunkSize()]);
87 // 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"
77 // The old resampler provides this memcpy facility in the case of matching
78 // sample rates, so reproduce it here for the sinc resampler.
|
| 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.
|
| /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/tests/ |
| README | 6 To build resampler library:
|
| resampler_tests.cpp | 41 android::AudioBufferProvider *provider, android::AudioResampler *resampler)
51 size_t framesResampled = resampler->resample(
95 // create the resampler
96 android::AudioResampler* resampler; local
98 resampler = android::AudioResampler::create(format, channels, outputFreq, quality);
99 resampler->setSampleRate(inputFreq);
100 resampler->setVolume(android::AudioResampler::UNITY_GAIN_FLOAT,
107 resample(channels, reference, outputFrames, refIncr, &provider, resampler);
113 resampler->reset();
115 delete resampler;
182 android::AudioResampler* resampler; local
[all...] |
| /external/webrtc/webrtc/common_audio/resampler/include/ |
| resampler.h | 26 class Resampler
30 Resampler();
31 Resampler(int inFreq, int outFreq, size_t num_channels);
32 ~Resampler();
89 Resampler* slave_left_;
90 Resampler* slave_right_;
|
| /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/webrtc/webrtc/common_audio/ |
| common_audio.gyp | 21 'resampler/include',
26 'resampler/include',
54 'resampler/include/push_resampler.h',
55 'resampler/include/resampler.h',
56 'resampler/push_resampler.cc',
57 'resampler/push_sinc_resampler.cc',
58 'resampler/push_sinc_resampler.h',
59 'resampler/resampler.cc'
[all...] |
| BUILD.gn | 14 "resampler/include",
44 "resampler/include/push_resampler.h",
45 "resampler/include/resampler.h",
46 "resampler/push_resampler.cc",
47 "resampler/push_sinc_resampler.cc",
48 "resampler/push_sinc_resampler.h",
49 "resampler/resampler.cc",
50 "resampler/sinc_resampler.cc"
[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...] |
| /frameworks/av/services/audioflinger/audio-resampler/ |
| Android.mk | 8 LOCAL_MODULE := libaudio-resampler
|
| /docs/source.android.com/src/devices/audio/ |
| src.jd | 40 another sample rate. A sample rate converter, or resampler, is a module
41 that implements sample rate conversion. With respect to the resampler,
50 internally. In that case, a resampler would be used to upsample the MP3
56 The characteristics of a resampler can be expressed using metrics, including:
63 <li>overall latency through the resampler</li>
72 The ideal resampler would exactly preserve the source signal's amplitude
85 Section <a href="#srcResamplers">Resampler implementations</a>
90 <h2 id="srcResamplers">Resampler implementations</h2>
93 Available resampler implementations change frequently,
114 The specific resampler implementation selected depends o
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| resample_input_audio_file.h | 17 #include "webrtc/common_audio/resampler/include/resampler.h"
45 Resampler resampler_;
|
| /external/webrtc/webrtc/voice_engine/ |
| utility.h | 18 #include "webrtc/common_audio/resampler/include/push_resampler.h"
35 PushResampler<int16_t>* resampler,
45 PushResampler<int16_t>* resampler,
|
| utility.cc | 14 #include "webrtc/common_audio/resampler/include/push_resampler.h"
25 PushResampler<int16_t>* resampler,
29 resampler, dst_frame);
39 PushResampler<int16_t>* resampler,
53 if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
63 int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
|
| /frameworks/av/services/audioflinger/ |
| test-resample.cpp | 49 fprintf(stderr," -q resampler quality\n");
343 AudioResampler* resampler = AudioResampler::create(format, channels, local
349 resampler->setSampleRate(9000);
350 resampler->setSampleRate(12000);
351 resampler->setSampleRate(20000);
352 resampler->setSampleRate(30000);
364 resampler->setSampleRate(1000);
368 resampler->setSampleRate(1000+i);
376 resampler->reset();
377 delete resampler;
381 AudioResampler* resampler = AudioResampler::create(format, channels, local
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| voice_activity_detector.h | 17 #include "webrtc/common_audio/resampler/include/resampler.h"
58 Resampler resampler_;
|
| /external/webrtc/webrtc/modules/utility/source/ |
| file_player_impl.h | 14 #include "webrtc/common_audio/resampler/include/resampler.h"
75 Resampler _resampler;
|
| file_recorder_impl.h | 21 #include "webrtc/common_audio/resampler/include/resampler.h"
90 Resampler _audioResampler;
|
| /device/htc/flounder/audio/hal/ |
| Android.mk | 10 # TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8
|
| /external/webrtc/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
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_resampler.cc | 16 #include "webrtc/common_audio/resampler/include/resampler.h"
|
| /device/asus/fugu/libaudio/ |
| AudioStreamIn.h | 21 #include <audio_utils/resampler.h>
81 // resampler buffer provider thunks
89 // resampler buffer provider methods
|
