| /sof-3.4.0/src/audio/pcm_converter/ |
| D | pcm_converter_hifi3.c | 43 ae_int16x4 sample = AE_ZERO16(); in pcm_convert_s16_to_s24() local
64 AE_LA16X4_IP(sample, inu, in); in pcm_convert_s16_to_s24()
66 AE_SA32X2_IP(AE_SRAI32(AE_CVT32X2F16_32(sample), 8), outu, out); in pcm_convert_s16_to_s24()
67 AE_SA32X2_IP(AE_SRAI32(AE_CVT32X2F16_10(sample), 8), outu, out); in pcm_convert_s16_to_s24()
76 AE_L16_IP(sample, (ae_int16 *)in, sizeof(ae_int16)); in pcm_convert_s16_to_s24()
78 /* shift right and store one 32 bit sample */ in pcm_convert_s16_to_s24()
79 AE_S32_L_IP(AE_SRAI32(AE_CVT32X2F16_32(sample), 8), (ae_int32 *)out, in pcm_convert_s16_to_s24()
92 * \param[in] sample Input sample.
93 * \return Shifted sample.
95 static ae_int32x2 pcm_shift_s24_to_s16(ae_int32x2 sample) in pcm_shift_s24_to_s16() argument
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| /sof-3.4.0/src/math/fft/ |
| D | fft_16_hifi3.c | 25 ae_int16x4 sample; in fft_execute_16() local
50 AE_L16_IP(sample, in, 0); in fft_execute_16()
51 sample = AE_NEG16S(sample); in fft_execute_16()
52 AE_S16_0_IP(sample, in, sizeof(struct icomplex16)); in fft_execute_16()
60 AE_L16_IP(sample, in, 2); in fft_execute_16()
61 sample = AE_SRAA16RS(sample, len); in fft_execute_16()
62 AE_S16_0_IP(sample, out, 2); in fft_execute_16()
64 AE_L16_IP(sample, in, 2); in fft_execute_16()
65 sample = AE_SRAA16RS(sample, len); in fft_execute_16()
66 AE_S16_0_IP(sample, out, 2); in fft_execute_16()
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| D | fft_32_hifi3.c | 24 ae_int32x2 sample; in fft_execute_32() local
49 AE_L32_IP(sample, in, 0); in fft_execute_32()
50 sample = AE_NEG32S(sample); in fft_execute_32()
51 AE_S32_L_IP(sample, in, sizeof(struct icomplex32)); in fft_execute_32()
58 AE_LA32X2_IP(sample, inu, inx); in fft_execute_32()
59 sample = AE_SRAA32S(sample, len); in fft_execute_32()
61 AE_SA32X2_IP(sample, outu, out); in fft_execute_32()
88 sample = AE_ROUND32X2F64SSYM(res, res1); in fft_execute_32()
92 sample2 = AE_ADD32S(sample1, sample); in fft_execute_32()
96 sample2 = AE_SUB32S(sample1, sample); in fft_execute_32()
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| /sof-3.4.0/src/audio/copier/ |
| D | copier_hifi.c | 30 ae_int32x2 sample; in apply_attenuation() local
41 comp_err(dev, "16bit sample isn't supported by attenuation"); in apply_attenuation()
53 AE_LA32X2_IP(sample, uu, in); in apply_attenuation()
54 sample = AE_SRAA32(sample, cd->attenuation); in apply_attenuation()
55 AE_SA32X2_IP(sample, su, out); in apply_attenuation()
59 AE_L32_IP(sample, (ae_int32 *)in, sizeof(ae_int32)); in apply_attenuation()
60 sample = AE_SRAA32(sample, cd->attenuation); in apply_attenuation()
61 AE_S32_L_IP(sample, (ae_int32 *)out, sizeof(ae_int32)); in apply_attenuation()
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| /sof-3.4.0/src/audio/mixer/ |
| D | mixer_hifi3.c | 22 ae_int16x4 sample = AE_ZERO16(); in mix_n_s16() local
50 AE_L16X4_IP(sample, in[j], 8); in mix_n_s16()
51 sample_1 = AE_SEXT32X2D16_32(sample); in mix_n_s16()
52 sample_2 = AE_SEXT32X2D16_10(sample); in mix_n_s16()
79 ae_int32x2 sample = AE_ZERO32(); in mix_n_s24() local
100 AE_L32X2_IP(sample, in[j], 8); in mix_n_s24()
102 sample = AE_SRAA32RS(AE_SLAI32(sample, 8), 8); in mix_n_s24()
103 val = AE_ADD32S(val, sample); in mix_n_s24()
123 ae_int64 sample; in mix_n_s32() local
147 /* load one 32 bit sample */ in mix_n_s32()
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| /sof-3.4.0/src/include/ipc/ |
| D | probe_dma_frame.h | 37 * D - 4 bits - Specify Sample Rate - value enumerating standard sample rates:
55 * F - 2 bits - Specify Sample Size, number of valid sample bytes minus 1
57 * H - 1 bit - Specifies Sample Format - 0 for Integer, 1 for Floating point
58 * I - 1 bit - Specifies Sample Endianness - 0 for LE
59 * J - 1 bit - Specifies Interleaving - 1 for Sample Interleaving
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| /sof-3.4.0/src/audio/drc/ |
| D | drc_hifi3.c | 99 int32_t sample; in drc_update_detector_average() local
115 if (nbyte == 2) { /* 2 bytes per sample */ in drc_update_detector_average()
120 sample = (int32_t)*sample16_p << 16; in drc_update_detector_average()
121 *abs_input_array_p = MAX(*abs_input_array_p, ABS(sample)); in drc_update_detector_average()
126 } else { /* 4 bytes per sample */ in drc_update_detector_average()
131 sample = *sample32_p; in drc_update_detector_average()
132 *abs_input_array_p = MAX(*abs_input_array_p, ABS(sample)); in drc_update_detector_average()
307 int32_t sample; in drc_compress_output() local
326 if (is_2byte) { /* 2 bytes per sample */ in drc_compress_output()
347 sample = (int32_t)*sample16_p; in drc_compress_output()
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| D | drc_generic.c | 87 int32_t sample; in drc_update_detector_average() local
102 if (nbyte == 2) { /* 2 bytes per sample */ in drc_update_detector_average()
108 sample = Q_SHIFT_LEFT((int32_t)*sample16_p, 15, 31); in drc_update_detector_average()
109 abs_input_array[i] = MAX(abs_input_array[i], ABS(sample)); in drc_update_detector_average()
112 } else { /* 4 bytes per sample */ in drc_update_detector_average()
118 sample = *sample32_p; in drc_update_detector_average()
119 abs_input_array[i] = MAX(abs_input_array[i], ABS(sample)); in drc_update_detector_average()
276 int32_t sample; in drc_compress_output() local
293 if (is_2byte) { /* 2 bytes per sample */ in drc_compress_output()
312 sample = (int32_t)*sample16_p; in drc_compress_output()
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| /sof-3.4.0/src/include/sof/audio/igo_nr/ |
| D | igo_nr_comp.h | 32 bool invalid_param; /**< sample rate != 16000 */
33 uint32_t sink_rate; /* Sample rate in Hz */
34 uint32_t source_rate; /* Sample rate in Hz */
35 uint32_t sink_format; /* For used PCM sample format */
36 uint32_t source_format; /* For used PCM sample format */
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| D | igo_lib.h | 69 uint16_t sample_num; /* Sample number in this data bulk */
123 * This API is used to process audio stream. The default audio sample is 16bit.
124 * The sampling rate and sample number should be specified in IgoStreamData
126 * interleaved sample by sample.
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| /sof-3.4.0/src/include/ipc4/ |
| D | asrc.h | 27 * ASRC_CONFIG Asynchronous Sample Rate Converter module configuration.
31 * - sample size (bit_depth): 32 bit
34 * Following sample conversion ratios are supported (input_frequency/output_frequency):
44 * - IBS calculated based on input frequency and sample group size
45 * - OBS calculated based on output frequency and sample group size, extended by X sample groups
48 * - IBS calculated based on input frequency and sample group size, extended by X sample groups
49 * - OBS calculated based on output frequency and sample group size
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| /sof-3.4.0/src/include/sof/audio/ |
| D | audio_stream.h | 58 enum sof_ipc_frame frame_fmt; /**< Sample data format */
81 * Retrieves readable address of a sample at specified index (see versions of
82 * this macro specialized for various sample types).
84 * @param idx Index of sample.
85 * @param size Size of sample in bytes.
86 * @return Pointer to the sample.
102 * Retrieves readable address of a signed 16-bit sample at specified index.
104 * @param idx Index of sample.
105 * @return Pointer to the sample.
113 * Retrieves readable address of a signed 32-bit sample at specified index.
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| D | pcm_converter.h | 42 * \param ioffset offset to first sample in source stream
44 * \param ooffset offset to first sample in sink stream
117 * \param valid_in_bits is source valid sample format.
119 * \param valid_out_bits is sink valid sample format.
159 * \param ioffset offset to first sample in source stream
161 * \param ooffset offset to first sample in sink stream
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| /sof-3.4.0/test/cmocka/src/audio/volume/ |
| D | volume_process.c | 123 int16_t sample; in verify_s16_to_s16() local
136 sample = (int16_t)processed; in verify_s16_to_s16()
137 delta = dst[i + channel] - sample; in verify_s16_to_s16()
139 assert_int_equal(dst[i + channel], sample); in verify_s16_to_s16()
170 int32_t sample; in verify_s24_to_s24_s32() local
188 sample = ((int32_t)processed) >> shift; in verify_s24_to_s24_s32()
190 delta = dst_sample - sample; in verify_s24_to_s24_s32()
192 assert_int_equal(dst_sample, sample); in verify_s24_to_s24_s32()
196 assert_int_equal(dst_sample, sample); in verify_s24_to_s24_s32()
227 int32_t sample; in verify_s32_to_s24_s32() local
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| /sof-3.4.0/src/audio/ |
| D | Kconfig | 131 need high sample rates.
149 so the full 20 kHz band is not met even if sample rate would
308 Select for Asynchronous sample rate conversion (ASRC)
338 for the asynchronous sample rate conversion. All the
347 In order to optimize the text code size of the sample rate
349 deactivated. Disregarding these settings, the sample rate
353 need an (asynchronous) 1:1 sample rate conversion, e.g, from
590 Support conversion between 16 bit valid sample size in 16 bit container
591 and 16 bit valid sample size in 32 bit container
597 Support conversion between 16 bit valid sample size in 32 bit container
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| /sof-3.4.0/src/include/sof/audio/asrc/ |
| D | asrc_farrow.h | 7 * @brief API header containing sample rate converter struct and interface
10 * @mainpage Hifi3 Implementation of the Intel TSD Sample Rate Converter
12 * The sample rate converter is based on the so-called Farrow structure.
16 * The sample rate converter can be applied for transmit and receive
17 * use cases. To support both directions, the sample rate converter
24 * If the sample rate converter operates in push-mode, the caller can
28 * the push-mode sample rate converter is usually combined with a ring
32 * If the sample rate converter operates in pull-mode, the caller can
36 * pull-mode sample rate converter is usually combined with a ring
66 * @brief Define whether the sample rate converter shall use a linear
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| /sof-3.4.0/src/audio/dcblock/ |
| D | dcblock_hifi4.c | 18 ae_int32x2 sample) in dcblock_cal() argument
24 out = AE_SUB64(AE_MOVAD32_L(sample), AE_MOVAD32_L(state_x)); in dcblock_cal()
53 ae_int32x2 R, state_x, state_y, sample; in dcblock_s16_default() local
67 /* Load a 16 bit sample*/ in dcblock_s16_default()
69 /* store the 16 bit sample to high 16bit of 32bit register*/ in dcblock_s16_default()
70 sample = AE_CVT32X2F16_32(in_sample); in dcblock_s16_default()
71 state_y = dcblock_cal(R, state_x, state_y, sample); in dcblock_s16_default()
72 state_x = sample; in dcblock_s16_default()
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| D | dcblock_hifi3.c | 18 ae_int32x2 sample) in dcblock_cal() argument
24 out = AE_SUB64(AE_MOVAD32_L(sample), AE_MOVAD32_L(state_x)); in dcblock_cal()
50 ae_int32x2 R, state_x, state_y, sample; in dcblock_s16_default() local
68 /* Load a 16 bit sample*/ in dcblock_s16_default()
70 /* store the 16 bit sample to high 16bit of 32bit register*/ in dcblock_s16_default()
71 sample = AE_CVT32X2F16_32(in_sample); in dcblock_s16_default()
72 state_y = dcblock_cal(R, state_x, state_y, sample); in dcblock_s16_default()
73 state_x = sample; in dcblock_s16_default()
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| /sof-3.4.0/src/samples/audio/ |
| D | kwd_nn_detect_test.c | 43 uint32_t sample; in kwd_nn_detect_test() local
48 for (sample = 0; sample < count && !test_keyword_get_detected(dev); ++sample) { in kwd_nn_detect_test()
50 audio_stream_read_frag_s16(source, sample) : in kwd_nn_detect_test()
51 audio_stream_read_frag_s32(source, sample); in kwd_nn_detect_test()
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| /sof-3.4.0/src/audio/aria/ |
| D | aria_hifi3.c | 35 /* maintains odd sample if any left */ in aria_algo_calc_gain()
84 /* variable for odd sample detection, detection when exceeded */ in aria_algo_get_data()
86 /* variable accumulates samples being processed, helps to identify odd sample */ in aria_algo_get_data()
102 /* below condition process channel pairs from current sample group in aria_algo_get_data()
115 /* below condition process odd channel from current sample group and next in aria_algo_get_data()
116 * sample group when channels count is odd, it process current in aria_algo_get_data()
117 * and next sample group corresponding to (idx) and (idx+1) in aria_algo_get_data()
132 /* maintains odd sample if any left */ in aria_algo_get_data()
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| /sof-3.4.0/src/audio/module_adapter/module/volume/ |
| D | volume_hifi3_with_peakvol.c | 101 /* process two continuous sample data once */ in vol_s24_to_s24_s32()
106 /* Load the input sample */ in vol_s24_to_s24_s32()
112 /* Multiply the input sample */ in vol_s24_to_s24_s32()
125 /* Store the output sample */ in vol_s24_to_s24_s32()
205 /* process two continuous sample data once */ in vol_s32_to_s24_s32()
210 /* Load the input sample */ in vol_s32_to_s24_s32()
328 /* Load the input sample */ in vol_s16_to_s16()
338 /* Multiply the input sample */ in vol_s16_to_s16()
413 /* set start address of sample load */ in vol_s24_to_s24_s32()
415 /* set start address of sample store */ in vol_s24_to_s24_s32()
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| /sof-3.4.0/app/ |
| D | sample.yaml | 1 sample:
9 sample.audio.sof:
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| /sof-3.4.0/src/drivers/intel/ |
| D | Kconfig | 75 number of channels, sample rate, and PCM format are
107 coefficients sets to support sample rates 8 - 96 kHz with
118 preserve support for 48 kHz and 16 kHz sample rates
148 Decimation by 3 in FIR is useful with microphone clock and sample
167 Decimation by 5 in FIR is useful with microphone clock and sample
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| /sof-3.4.0/tools/test/audio/ |
| D | sof_test_perf_config.m | 15 play.sft = 'S16_LE'; % Sample format to use
26 rec.sft = 'S24_3LE'; % Sample format to use
32 test.fs = 48000; % Sample rate
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| /sof-3.4.0/tools/tune/src/ |
| D | src_factor1_lm.m | 3 % factor1_lm - factorize input and output sample rates ratio to fraction l/m
7 % fs1 - input sample rate
8 % fs2 - output sample rate
|
