| /sof-2.7.6/src/audio/pcm_converter/ |
| D | pcm_converter_hifi3.c | 50 uint32_t ooffset, uint32_t samples) in pcm_convert_s16_to_s24() argument
61 if (!samples) in pcm_convert_s16_to_s24()
62 return samples; in pcm_convert_s16_to_s24()
79 if (++i == samples) in pcm_convert_s16_to_s24()
80 return samples; in pcm_convert_s16_to_s24()
87 while (samples >= 3 && i < samples - 3) { in pcm_convert_s16_to_s24()
112 while (i++ != samples) { in pcm_convert_s16_to_s24()
127 return samples; in pcm_convert_s16_to_s24()
155 uint32_t ooffset, uint32_t samples) in pcm_convert_s24_to_s16() argument
169 if (!samples) in pcm_convert_s24_to_s16()
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| D | pcm_converter_generic.c | 32 uint32_t ooffset, uint32_t samples) in pcm_convert_s16_to_s24() argument
39 for (i = 0; i < samples; i++) { in pcm_convert_s16_to_s24()
46 return samples; in pcm_convert_s16_to_s24()
51 uint32_t ooffset, uint32_t samples) in pcm_convert_s24_to_s16() argument
58 for (i = 0; i < samples; i++) { in pcm_convert_s24_to_s16()
65 return samples; in pcm_convert_s24_to_s16()
74 uint32_t ooffset, uint32_t samples) in pcm_convert_s16_to_s32() argument
81 for (i = 0; i < samples; i++) { in pcm_convert_s16_to_s32()
88 return samples; in pcm_convert_s16_to_s32()
93 uint32_t ooffset, uint32_t samples) in pcm_convert_s32_to_s16() argument
[all …]
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| D | pcm_converter.c | 19 uint32_t samples, pcm_converter_lin_func converter) in pcm_convert_as_linear() argument
34 if (audio_stream_get_avail_samples(source) < samples + ioffset) in pcm_convert_as_linear()
36 if (audio_stream_get_free_samples(sink) < samples + ooffset) in pcm_convert_as_linear()
39 while (i < samples) { in pcm_convert_as_linear()
47 chunk = MIN(chunk, samples - i); in pcm_convert_as_linear()
58 return samples; in pcm_convert_as_linear()
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| /sof-2.7.6/src/include/sof/audio/ |
| D | pcm_converter.h | 47 uint32_t ooffset, uint32_t samples);
56 uint32_t samples);
153 uint32_t samples, pcm_converter_lin_func converter);
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| D | audio_stream.h | 558 uint32_t ooffset, uint32_t samples) in audio_stream_copy() argument
565 uint32_t bytes = samples * ssize; in audio_stream_copy()
587 return samples; in audio_stream_copy()
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| /sof-2.7.6/src/lib/ |
| D | dma.c | 189 uint32_t samples = source_bytes / in dma_buffer_copy_from() local
192 samples; in dma_buffer_copy_from()
199 ret = process(istream, 0, &sink->stream, 0, samples); in dma_buffer_copy_from()
217 uint32_t samples = sink_bytes / in dma_buffer_copy_to() local
220 samples; in dma_buffer_copy_to()
226 ret = process(&source->stream, 0, ostream, 0, samples); in dma_buffer_copy_to()
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| /sof-2.7.6/tools/test/audio/test_utils/ |
| D | load_test_input.m | 13 % x - samples
14 % n - number of samples
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| D | find_test_signal.m | 11 % nt - number of samples in test tone
12 % nt_use - number of samples to use
13 % nt_skip - number of samples to skip
63 %% Delay to first tone, length of tone in samples
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| D | load_test_output.m | 13 % x - samples
14 % n - number of samples
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| D | mix_sweep.m | 7 %% Adjust tone lengt to integer number of samples
8 test.nt = round(test.tl*test.fs); % Make number of samples per tone
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| D | mix_chirp.m | 33 %% Adjust tone length to integer number of samples
34 test.nt = round(test.cl*test.fs); % Make number of samples per tone
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| D | chirp_test_analyze.m | 34 % Check for proper ratio of out/in samples, minimum level, maximum offset
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| /sof-2.7.6/src/samples/ |
| D | Kconfig | 6 bool "Build samples"
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| /sof-2.7.6/src/ |
| D | Kconfig | 15 rsource "samples/Kconfig"
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| D | CMakeLists.txt | 10 add_subdirectory(samples)
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| /sof-2.7.6/test/cmocka/src/audio/mixer/ |
| D | mixer_test.c | 272 uint32_t *samples = tc->sources[src_idx].buf->stream.addr; in test_audio_mixer_copy() local
277 samples[smp] = ((sin(rad) + 1) / 2) * (0xFFFFFFFF / 2); in test_audio_mixer_copy()
292 uint32_t *samples = in test_audio_mixer_copy() local
295 sum += samples[smp]; in test_audio_mixer_copy()
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| /sof-2.7.6/src/samples/audio/ |
| D | Kconfig | 3 menu "Audio component samples"
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| /sof-2.7.6/src/audio/ |
| D | google_hotword_detect.c | 333 const void *samples, in ghd_detect() argument
357 (uint32_t)samples, bytes / sample_bytes); in ghd_detect()
358 ret = GoogleHotwordDspProcess(samples, bytes / sample_bytes, in ghd_detect()
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| D | dai.c | 906 uint32_t samples; in dai_copy() local
925 samples = MIN(src_samples, sink_samples); in dai_copy()
929 samples = MIN(src_samples, sink_samples); in dai_copy()
935 samples = MIN(samples, dd->period_bytes / sampling); in dai_copy()
937 copy_bytes = samples * sampling; in dai_copy()
943 samples / buf->stream.channels); in dai_copy()
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| /sof-2.7.6/tools/topology/topology1/platform/intel/ |
| D | cml.m4 | 42 # for sof-cml-rt5682-max98357.m4 uses 3 parameters and bits per samples set by SSP1_VALID_BITS
65 # for sof-cml-rt5682-max98357.m4 uses 3 parameters and bits per samples set by SSP1_VALID_BITS
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| /sof-2.7.6/test/cmocka/src/audio/pcm_converter/ |
| D | pcm_float.c | 73 int samples, const void *data) in _test_pcm_convert() argument
80 const int inbytes = samples * get_sample_bytes(frm_in); in _test_pcm_convert()
81 const int outbytes = (samples + 1) * get_sample_bytes(frm_out); in _test_pcm_convert()
97 fun(&source->stream, 0, &sink->stream, 0, samples); in _test_pcm_convert()
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| /sof-2.7.6/tools/test/audio/std_utils/ |
| D | g_test_input.m | 67 % 0.5 seconds tone, this will be adjusted to be integer number of samples
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| /sof-2.7.6/zephyr/ |
| D | README | 48 % west build -p always -b up_squared_adsp samples/basic/minimal
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| /sof-2.7.6/tools/tune/crossover/ |
| D | crossover_plot_freq.m | 3 % that represents the path the samples go through for each sinks.
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| /sof-2.7.6/tools/tune/tdfb/ |
| D | bf_design.m | 376 nt = bf.fs * bf.sinerot_t; % Number samples output per angle
377 nti = p * nt; % Number samples output per angle at high rate
417 nt = floor(bf.fs * bf.diffuse_t); % Number samples output per angle
418 nti = p * nt; % Number samples output per angle at high rate
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