| /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_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.
|
| 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/audio-resampler/ |
| Android.mk | 8 LOCAL_MODULE := libaudio-resampler
|
| /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/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',
|
| /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',
|
| /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_;
|
| /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_;
|
| /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,
|
| /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/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/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_opus.h | 14 #include "webrtc/common_audio/resampler/include/resampler.h"
|
| acm_resampler.cc | 16 #include "webrtc/common_audio/resampler/include/resampler.h"
|
