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/sof-3.4.0/src/audio/selector/
Dselector_generic.c36 const struct audio_stream __sparse_cache *source, uint32_t frames) in sel_s16le_1ch() argument
39 int16_t *src = source->r_ptr; in sel_s16le_1ch()
46 const int source_frame_bytes = audio_stream_frame_bytes(source); in sel_s16le_1ch()
47 const unsigned int nch = source->channels; in sel_s16le_1ch()
52 nmax = audio_stream_bytes_without_wrap(source, src) / source_frame_bytes; in sel_s16le_1ch()
62 src = audio_stream_wrap(source, src + nch * n); in sel_s16le_1ch()
76 const struct audio_stream __sparse_cache *source, uint32_t frames) in sel_s16le_nch() argument
78 int8_t *src = (int8_t *)source->r_ptr; in sel_s16le_nch()
83 const int bytes_total = frames * audio_stream_frame_bytes(source); in sel_s16le_nch()
87 bmax = audio_stream_bytes_without_wrap(source, src); in sel_s16le_nch()
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/sof-3.4.0/tools/tplg_parser/
Dgraph.c29 char *source = NULL, *sink = NULL; in tplg_create_graph() local
46 graph_elem->source) == 0) { in tplg_create_graph()
48 source = graph_elem->source; in tplg_create_graph()
58 if (!source || !sink) { in tplg_create_graph()
60 __func__, source, sink); in tplg_create_graph()
64 printf("loading route %s -> %s\n", source, sink); in tplg_create_graph()
66 strcat(pipeline_string, graph_elem->source); in tplg_create_graph()
/sof-3.4.0/src/audio/pcm_converter/
Dpcm_converter_generic.c34 static int pcm_convert_s16_to_s24(const struct audio_stream __sparse_cache *source, in pcm_convert_s16_to_s24() argument
38 int16_t *src = source->r_ptr; in pcm_convert_s16_to_s24()
46 src = audio_stream_wrap(source, src); in pcm_convert_s16_to_s24()
49 nmax = audio_stream_bytes_without_wrap(source, src) >> BYTES_TO_S16_SAMPLES; in pcm_convert_s16_to_s24()
63 static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *source, in pcm_convert_s24_to_s16() argument
67 int32_t *src = source->r_ptr; in pcm_convert_s24_to_s16()
75 src = audio_stream_wrap(source, src); in pcm_convert_s24_to_s16()
78 nmax = audio_stream_bytes_without_wrap(source, src) >> BYTES_TO_S32_SAMPLES; in pcm_convert_s24_to_s16()
96 static int pcm_convert_s16_to_s32(const struct audio_stream __sparse_cache *source, in pcm_convert_s16_to_s32() argument
100 int16_t *src = source->r_ptr; in pcm_convert_s16_to_s32()
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Dpcm_converter.c18 int pcm_convert_as_linear(const struct audio_stream __sparse_cache *source, uint32_t ioffset, in pcm_convert_as_linear() argument
22 const int s_size_in = audio_stream_sample_bytes(source); in pcm_convert_as_linear()
26 char *r_ptr = audio_stream_get_frag(source, source->r_ptr, ioffset, in pcm_convert_as_linear()
35 if (audio_stream_get_avail_samples(source) < samples + ioffset) in pcm_convert_as_linear()
43 N1 = audio_stream_bytes_without_wrap(source, r_ptr) >> in pcm_convert_as_linear()
54 r_ptr = audio_stream_wrap(source, r_ptr + chunk * s_size_in); in pcm_convert_as_linear()
Dpcm_converter_hifi3.c39 static int pcm_convert_s16_to_s24(const struct audio_stream __sparse_cache *source, in pcm_convert_s16_to_s24() argument
47 int16_t *src = source->r_ptr; in pcm_convert_s16_to_s24()
50 ae_int16x4 *in = audio_stream_wrap(source, src + ioffset); in pcm_convert_s16_to_s24()
54 nmax = audio_stream_samples_without_wrap_s16(source, in); in pcm_convert_s16_to_s24()
83 in = audio_stream_wrap(source, in); in pcm_convert_s16_to_s24()
113 static int pcm_convert_s24_to_s16(const struct audio_stream __sparse_cache *source, in pcm_convert_s24_to_s16() argument
123 ae_int32 *src = source->r_ptr; in pcm_convert_s24_to_s16()
126 ae_int32x2 *in = audio_stream_wrap(source, src + ioffset); in pcm_convert_s24_to_s16()
130 nmax = audio_stream_samples_without_wrap_s32(source, in); in pcm_convert_s24_to_s16()
167 in = audio_stream_wrap(source, in); in pcm_convert_s24_to_s16()
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/sof-3.4.0/src/audio/module_adapter/module/volume/
Dvolume_generic.c64 struct audio_stream __sparse_cache *source = bsource->data; in vol_s24_to_s24() local
70 const int nch = source->channels; in vol_s24_to_s24()
73 x = audio_stream_wrap(source, (char *)source->r_ptr + bsource->consumed); in vol_s24_to_s24()
79 nmax = audio_stream_samples_without_wrap_s24(source, x); in vol_s24_to_s24()
92 x = audio_stream_wrap(source, x + n); in vol_s24_to_s24()
115 struct audio_stream __sparse_cache *source = bsource->data; in vol_s32_to_s32() local
121 const int nch = source->channels; in vol_s32_to_s32()
124 x = audio_stream_wrap(source, (char *)source->r_ptr + bsource->consumed); in vol_s32_to_s32()
129 nmax = audio_stream_samples_without_wrap_s32(source, x); in vol_s32_to_s32()
146 x = audio_stream_wrap(source, x + n); in vol_s32_to_s32()
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Dvolume_generic_with_peakvol.c60 struct audio_stream __sparse_cache *source = bsource->data; in vol_s24_to_s24() local
66 const int nch = source->channels; in vol_s24_to_s24()
71 x = audio_stream_wrap(source, (char *)source->r_ptr + bsource->consumed); in vol_s24_to_s24()
77 nmax = audio_stream_samples_without_wrap_s24(source, x); in vol_s24_to_s24()
94 x = audio_stream_wrap(source, x + n); in vol_s24_to_s24()
119 struct audio_stream __sparse_cache *source = bsource->data; in vol_s32_to_s32() local
125 const int nch = source->channels; in vol_s32_to_s32()
130 x = audio_stream_wrap(source, (char *)source->r_ptr + bsource->consumed); in vol_s32_to_s32()
135 nmax = audio_stream_samples_without_wrap_s32(source, x); in vol_s32_to_s32()
156 x = audio_stream_wrap(source, x + n); in vol_s32_to_s32()
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/sof-3.4.0/test/cmocka/src/audio/selector/
Dselector_test.c25 struct comp_buffer *source; member
27 struct audio_stream *source);
39 struct audio_stream *source);
83 sel_state->source = create_test_source(sel_state->dev, 0, parameters->source_format, in setup()
104 free_test_source(sel_state->source); in teardown()
113 struct comp_buffer *source; member
119 struct audio_stream *source);
131 struct audio_stream *source);
182 sel_state->source = create_test_source(dev, 0, parameters->source_format, in setup()
186 sel_state->input->data = &sel_state->source->stream; in setup()
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/sof-3.4.0/src/audio/smart_amp/
Dsmart_amp_generic.c27 const struct audio_stream __sparse_cache *source, in smart_amp_s16_ff_default() argument
38 int nch = source->channels; in smart_amp_s16_ff_default()
43 x = audio_stream_read_frag_s16(source, idx); in smart_amp_s16_ff_default()
55 const struct audio_stream __sparse_cache *source, in smart_amp_s24_ff_default() argument
66 int nch = source->channels; in smart_amp_s24_ff_default()
71 x = audio_stream_read_frag_s32(source, idx); in smart_amp_s24_ff_default()
83 const struct audio_stream __sparse_cache *source, in smart_amp_s32_ff_default() argument
93 int nch = source->channels; in smart_amp_s32_ff_default()
98 x = audio_stream_read_frag_s32(source, idx); in smart_amp_s32_ff_default()
109 const struct audio_stream __sparse_cache *source, in smart_amp_s16_fb_default() argument
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/sof-3.4.0/src/audio/eq_fir/
Deq_fir_generic.c26 struct audio_stream __sparse_cache *source = bsource->data; in eq_fir_s16() local
31 int16_t *x = source->r_ptr; in eq_fir_s16()
34 int nch = source->channels; in eq_fir_s16()
38 nmax = EQ_FIR_BYTES_TO_S16_SAMPLES(audio_stream_bytes_without_wrap(source, x)); in eq_fir_s16()
54 x = audio_stream_wrap(source, x + n); in eq_fir_s16()
64 struct audio_stream __sparse_cache *source = bsource->data; in eq_fir_s24() local
69 int32_t *x = source->r_ptr; in eq_fir_s24()
72 int nch = source->channels; in eq_fir_s24()
76 nmax = EQ_FIR_BYTES_TO_S32_SAMPLES(audio_stream_bytes_without_wrap(source, x)); in eq_fir_s24()
92 x = audio_stream_wrap(source, x + n); in eq_fir_s24()
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Deq_fir_hifi3.c30 struct audio_stream __sparse_cache *source = bsource->data; in eq_fir_2x_s32() local
35 ae_int32 *src = (ae_int32 *)source->r_ptr; in eq_fir_2x_s32()
45 int nch = source->channels; in eq_fir_2x_s32()
57 fir_comp_setup_circular(source); in eq_fir_2x_s32()
72 fir_comp_setup_circular(source); in eq_fir_2x_s32()
93 struct audio_stream __sparse_cache *source = bsource->data; in eq_fir_2x_s24() local
100 ae_int32 *src = (ae_int32 *)source->r_ptr; in eq_fir_2x_s24()
109 int nch = source->channels; in eq_fir_2x_s24()
120 fir_comp_setup_circular(source); in eq_fir_2x_s24()
133 fir_comp_setup_circular(source); in eq_fir_2x_s24()
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/sof-3.4.0/src/include/sof/audio/
Daudio_stream.h408 static inline int audio_stream_can_copy_bytes(const struct audio_stream __sparse_cache *source, in audio_stream_can_copy_bytes() argument
413 if (audio_stream_get_avail_bytes(source) < bytes) in audio_stream_can_copy_bytes()
433 audio_stream_get_copy_bytes(const struct audio_stream __sparse_cache *source, in audio_stream_get_copy_bytes() argument
436 uint32_t avail = audio_stream_get_avail_bytes(source); in audio_stream_get_copy_bytes()
454 audio_stream_avail_frames(const struct audio_stream __sparse_cache *source, in audio_stream_avail_frames() argument
457 uint32_t src_frames = audio_stream_get_avail_frames(source); in audio_stream_avail_frames()
472 audio_stream_avail_frames_aligned(const struct audio_stream __sparse_cache *source, in audio_stream_avail_frames_aligned() argument
475 uint32_t src_frames = (audio_stream_get_avail_bytes(source) >> source->frame_align_shift) in audio_stream_avail_frames_aligned()
476 * source->frame_align; in audio_stream_avail_frames_aligned()
622 audio_stream_bytes_without_wrap(const struct audio_stream __sparse_cache *source, in audio_stream_bytes_without_wrap() argument
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Dpcm_converter.h48 typedef int (*pcm_converter_func)(const struct audio_stream __sparse_cache *source,
63 enum sof_ipc_frame source; /**< source frame format */ member
86 if (in != pcm_func_map[i].source) in pcm_get_conversion_function()
99 enum sof_ipc_frame source; /**< source frame container format */ member
134 if (in_bits != pcm_func_vc_map[i].source) in pcm_get_conversion_vc_function()
166 int pcm_convert_as_linear(const struct audio_stream __sparse_cache *source, uint32_t ioffset,
/sof-3.4.0/src/audio/
Dcomponent.c161 void comp_get_copy_limits(struct comp_buffer __sparse_cache *source, in comp_get_copy_limits() argument
165 cl->frames = audio_stream_avail_frames(&source->stream, &sink->stream); in comp_get_copy_limits()
166 cl->source_frame_bytes = audio_stream_frame_bytes(&source->stream); in comp_get_copy_limits()
172 void comp_get_copy_limits_frame_aligned(const struct comp_buffer __sparse_cache *source, in comp_get_copy_limits_frame_aligned() argument
176 cl->frames = audio_stream_avail_frames_aligned(&source->stream, &sink->stream); in comp_get_copy_limits_frame_aligned()
177 cl->source_frame_bytes = audio_stream_frame_bytes(&source->stream); in comp_get_copy_limits_frame_aligned()
185 int audio_stream_copy(const struct audio_stream __sparse_cache *source, uint32_t ioffset, in audio_stream_copy() argument
188 int ssize = audio_stream_sample_bytes(source); /* src fmt == sink fmt */ in audio_stream_copy()
189 ae_int16x4 *src = (ae_int16x4 *)((int8_t *)source->r_ptr + ioffset * ssize); in audio_stream_copy()
202 src = audio_stream_wrap(source, src); in audio_stream_copy()
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/sof-3.4.0/src/audio/dcblock/
Ddcblock_generic.c39 const struct audio_stream __sparse_cache *source, in dcblock_s16_default() argument
45 int16_t *x = source->r_ptr; in dcblock_s16_default()
52 int nch = source->channels; in dcblock_s16_default()
56 nmax = audio_stream_samples_without_wrap_s16(source, x); in dcblock_s16_default()
71 x = audio_stream_wrap(source, x + n); in dcblock_s16_default()
80 const struct audio_stream __sparse_cache *source, in dcblock_s24_default() argument
86 int32_t *x = source->r_ptr; in dcblock_s24_default()
93 int nch = source->channels; in dcblock_s24_default()
97 nmax = audio_stream_samples_without_wrap_s24(source, x); in dcblock_s24_default()
112 x = audio_stream_wrap(source, x + n); in dcblock_s24_default()
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Ddcblock_hifi3.c32 static inline void dcblock_set_circular(const struct audio_stream __sparse_cache *source) in dcblock_set_circular() argument
35 AE_SETCBEGIN0(source->addr); in dcblock_set_circular()
36 AE_SETCEND0(source->end_addr); in dcblock_set_circular()
41 const struct audio_stream __sparse_cache *source, in dcblock_s16_default() argument
46 ae_int16 *src = (ae_int16 *)source->r_ptr; in dcblock_s16_default()
53 int nch = source->channels; in dcblock_s16_default()
57 dcblock_set_circular(source); in dcblock_s16_default()
82 src = audio_stream_wrap(source, src + n); in dcblock_s16_default()
89 const struct audio_stream __sparse_cache *source, in dcblock_s24_default() argument
94 ae_int32 *src = (ae_int32 *)source->r_ptr; in dcblock_s24_default()
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Ddcblock_hifi4.c32 static inline void dcblock_set_circular(const struct audio_stream __sparse_cache *source, in dcblock_set_circular() argument
36 AE_SETCBEGIN0(source->addr); in dcblock_set_circular()
37 AE_SETCEND0(source->end_addr); in dcblock_set_circular()
46 const struct audio_stream __sparse_cache *source, in dcblock_s16_default() argument
56 int nch = source->channels; in dcblock_s16_default()
59 dcblock_set_circular(source, sink); in dcblock_s16_default()
61 in = (ae_int16 *)source->r_ptr + ch; in dcblock_s16_default()
84 const struct audio_stream __sparse_cache *source, in dcblock_s24_default() argument
94 int nch = source->channels; in dcblock_s24_default()
97 dcblock_set_circular(source, sink); in dcblock_s24_default()
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/sof-3.4.0/src/audio/mixin_mixout/
Dmixin_mixout_generic.c22 const struct audio_stream __sparse_cache *source, in normal_mix_channel_s16() argument
30 int16_t *src = (int16_t *)source->r_ptr; in normal_mix_channel_s16()
38 src = audio_stream_wrap(source, src); in normal_mix_channel_s16()
41 nmax = audio_stream_samples_without_wrap_s16(source, src); in normal_mix_channel_s16()
52 src = audio_stream_wrap(source, src); in normal_mix_channel_s16()
54 nmax = audio_stream_samples_without_wrap_s16(source, src); in normal_mix_channel_s16()
65 const struct audio_stream __sparse_cache *source, in remap_mix_channel_s16() argument
75 src = (int16_t *)source->r_ptr + source_channel_index; in remap_mix_channel_s16()
83 src = audio_stream_wrap(source, src); in remap_mix_channel_s16()
86 nmax = audio_stream_samples_without_wrap_s16(source, src); in remap_mix_channel_s16()
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/sof-3.4.0/src/audio/mux/
Dmux_generic.c26 const struct audio_stream __sparse_cache *source; in mux_check_for_wrap() local
31 source = sources[lookup->copy_elem[elem].stream_id]; in mux_check_for_wrap()
35 audio_stream_wrap(source, lookup->copy_elem[elem].src); in mux_check_for_wrap()
40 const struct audio_stream __sparse_cache *source, in demux_check_for_wrap() argument
50 audio_stream_wrap(source, lookup->copy_elem[elem].src); in demux_check_for_wrap()
58 __sparse_cache *source, in demux_calc_frames_without_wrap_s16() argument
75 frames = audio_stream_frames_without_wrap(source, ptr); in demux_calc_frames_without_wrap_s16()
87 const struct audio_stream __sparse_cache *source; in mux_calc_frames_without_wrap_s16() local
102 source = sources[lookup->copy_elem[elem].stream_id]; in mux_calc_frames_without_wrap_s16()
106 frames = audio_stream_frames_without_wrap(source, ptr); in mux_calc_frames_without_wrap_s16()
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/sof-3.4.0/src/audio/tdfb/
Dtdfb_generic.c61 struct audio_stream __sparse_cache *source = bsource->data; in tdfb_fir_s16() local
63 int16_t *x = source->r_ptr; in tdfb_fir_s16()
69 const int in_nch = source->channels; in tdfb_fir_s16()
75 fmax = audio_stream_frames_without_wrap(source, x); in tdfb_fir_s16()
97 x = audio_stream_wrap(source, x); in tdfb_fir_s16()
107 struct audio_stream __sparse_cache *source = bsource->data; in tdfb_fir_s24() local
109 int32_t *x = source->r_ptr; in tdfb_fir_s24()
115 const int in_nch = source->channels; in tdfb_fir_s24()
121 fmax = audio_stream_frames_without_wrap(source, x); in tdfb_fir_s24()
143 x = audio_stream_wrap(source, x); in tdfb_fir_s24()
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Dtdfb_hifiep.c61 struct audio_stream __sparse_cache *source = bsource->data; in tdfb_fir_s16() local
63 int16_t *x = source->r_ptr; in tdfb_fir_s16()
69 const int in_nch = source->channels; in tdfb_fir_s16()
75 fmax = audio_stream_frames_without_wrap(source, x); in tdfb_fir_s16()
97 x = audio_stream_wrap(source, x); in tdfb_fir_s16()
107 struct audio_stream __sparse_cache *source = bsource->data; in tdfb_fir_s24() local
109 int32_t *x = source->r_ptr; in tdfb_fir_s24()
115 const int in_nch = source->channels; in tdfb_fir_s24()
121 fmax = audio_stream_frames_without_wrap(source, x); in tdfb_fir_s24()
143 x = audio_stream_wrap(source, x); in tdfb_fir_s24()
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/sof-3.4.0/src/audio/mixer/
Dmixer.c174 struct comp_buffer *source = container_of(blist, struct comp_buffer, in mixer_reset() local
176 struct comp_buffer __sparse_cache *source_c = buffer_acquire(source); in mixer_reset()
179 if (source_c->source && source_c->source->state > COMP_STATE_READY) in mixer_reset()
196 static inline void mixer_set_frame_alignment(struct audio_stream __sparse_cache *source) in mixer_set_frame_alignment() argument
203 const uint32_t byte_align = source->channels == 6 ? 16 : 8; in mixer_set_frame_alignment()
218 audio_stream_init_alignment_constants(byte_align, frame_align_req, source); in mixer_set_frame_alignment()
238 struct comp_buffer *source; in mixer_prepare() local
250 source = container_of(blist, struct comp_buffer, sink_list); in mixer_prepare()
251 source_c = buffer_acquire(source); in mixer_prepare()
253 stop = source_c->source && (source_c->source->state == COMP_STATE_PAUSED || in mixer_prepare()
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/sof-3.4.0/src/audio/eq_iir/
Deq_iir.c67 struct audio_stream __sparse_cache *source = bsource->data; in eq_iir_s16_default() local
80 const int nch = source->channels; in eq_iir_s16_default()
84 x = source->r_ptr; in eq_iir_s16_default()
88 n1 = audio_stream_bytes_without_wrap(source, x) >> 1; in eq_iir_s16_default()
103 x = audio_stream_wrap(source, x + n); in eq_iir_s16_default()
115 struct audio_stream __sparse_cache *source = bsource->data; in eq_iir_s24_default() local
128 const int nch = source->channels; in eq_iir_s24_default()
132 x = source->r_ptr; in eq_iir_s24_default()
136 n1 = audio_stream_bytes_without_wrap(source, x) >> 2; in eq_iir_s24_default()
151 x = audio_stream_wrap(source, x + n); in eq_iir_s24_default()
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/sof-3.4.0/test/cmocka/src/math/fft/
Dfft.c268 struct comp_buffer *source = buffer_new(&test_buf_desc); in test_math_fft_256() local
271 int32_t *in = (int32_t *)source->stream.addr; in test_math_fft_256()
283 source->stream.channels = 1; in test_math_fft_256()
286 fft_real(source, sink, fft_size); in test_math_fft_256()
310 struct comp_buffer *source = buffer_new(&test_buf_desc); in test_math_fft_512() local
313 int32_t *in = (int32_t *)source->stream.addr; in test_math_fft_512()
325 source->stream.channels = 1; in test_math_fft_512()
328 fft_real(source, sink, fft_size); in test_math_fft_512()
352 struct comp_buffer *source = buffer_new(&test_buf_desc); in test_math_fft_1024() local
355 int32_t *in = (int32_t *)source->stream.addr; in test_math_fft_1024()
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/sof-3.4.0/tools/testbench/
Dfile.c60 static void sign_extend_source_s24(const struct audio_stream *source, int samples) in sign_extend_source_s24() argument
63 int32_t *src = (int32_t *)source->r_ptr; in sign_extend_source_s24()
70 bytes_src = audio_stream_bytes_without_wrap(source, src); in sign_extend_source_s24()
78 src = audio_stream_wrap(source, src); in sign_extend_source_s24()
114 static int write_binary_s32(struct file_comp_data *cd, const struct audio_stream *source, in write_binary_s32() argument
117 int32_t *src = (int32_t *)source->r_ptr; in write_binary_s32()
125 bytes_src = audio_stream_bytes_without_wrap(source, src); in write_binary_s32()
134 src = audio_stream_wrap(source, src + ret); in write_binary_s32()
173 static int write_text_s32(struct file_comp_data *cd, const struct audio_stream *source, int samples) in write_text_s32() argument
175 int32_t *src = (int32_t *)source->r_ptr; in write_text_s32()
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