| /third_party/ffmpeg/libavresample/x86/ |
| D | w64xmmtest.c | 26 int out_plane_size, int out_samples, uint8_t **input, in wrap() argument
30 out_samples, input, in_plane_size, in_samples); in wrap()
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| /third_party/ffmpeg/libavresample/aarch64/ |
| D | neontest.c | 26 int out_plane_size, int out_samples, uint8_t **input, in wrap() argument
30 out_samples, input, in_plane_size, in_samples); in wrap()
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| /third_party/ffmpeg/libavresample/arm/ |
| D | neontest.c | 26 int out_plane_size, int out_samples, uint8_t **input, in wrap() argument
30 out_samples, input, in_plane_size, in_samples); in wrap()
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| /third_party/ffmpeg/libavresample/ |
| D | resample.c | 336 int ch, in_samples, in_leftover, consumed = 0, out_samples = 0; in ff_audio_resample() local
406 out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples, in ff_audio_resample()
408 ret = ff_audio_data_realloc(dst, out_samples); in ff_audio_resample()
417 out_samples = resample(c, (void *)dst->data[ch], in ff_audio_resample()
422 if (out_samples < 0) { in ff_audio_resample()
424 return out_samples; in ff_audio_resample()
432 in_samples, in_leftover, out_samples, c->buffer->nb_samples); in ff_audio_resample()
434 dst->nb_samples = out_samples; in ff_audio_resample()
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| D | utils.c | 332 int out_samples, in avresample_convert() argument
364 avr->out_channels, out_samples, in avresample_convert()
381 !avr->out_convert_needed && direct_output && out_samples >= in_samples) { in avresample_convert()
392 (!direct_output || out_samples < in_samples)) { in avresample_convert()
443 if (!avr->out_convert_needed && direct_output && out_samples > 0) in avresample_convert()
478 if (direct_output && out_samples >= current_buffer->nb_samples) { in avresample_convert()
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| D | avresample.h | 452 int out_plane_size, int out_samples,
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| /third_party/gstreamer/gstplugins_base/gst-libs/gst/audio/ |
| D | gstaudioringbuffer.c | 54 guint8 * data, gint in_samples, gint out_samples, gint * accum);
1438 out_samples -= towrite / bpf; \
1458 out_samples -= (d - db)/bpf; \
1477 out_samples -= (d - db)/bpf; \
1495 out_samples -= (d - db)/bpf; \
1513 out_samples -= (d - db)/bpf; \
1519 guint8 * data, gint in_samples, gint out_samples, gint * accum) in default_commit() argument
1543 reverse = out_samples < 0; in default_commit()
1544 out_samples = ABS (out_samples); in default_commit()
1546 if (in_samples >= out_samples) in default_commit()
[all …]
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| D | gstaudioringbuffer.h | 265 gint out_samples, gint * accum);
378 gint out_samples, gint * accum);
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| D | gstaudiobasesink.c | 1815 gint out_samples; in gst_audio_base_sink_render() local
2130 out_samples = render_stop - render_start; in gst_audio_base_sink_render()
2139 sample_offset, samples, out_samples); in gst_audio_base_sink_render()
2148 info.data + offset, samples, out_samples, &accum); in gst_audio_base_sink_render()
2165 if (out_samples > written) { in gst_audio_base_sink_render()
2166 out_samples -= written; in gst_audio_base_sink_render()
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| /third_party/ffmpeg/libavcodec/ |
| D | atrac3.c | 644 float **out_samples) in decode_frame() argument
670 out_samples[ch], ch, JOINT_STEREO); in decode_frame()
715 out_samples[ch+1], ch+1, JOINT_STEREO); in decode_frame()
720 reverse_matrixing(out_samples[ch], out_samples[ch+1], in decode_frame()
724 channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]); in decode_frame()
736 out_samples[i], i, q->coding_mode); in decode_frame()
744 float *p1 = out_samples[i]; in decode_frame()
757 int size, float **out_samples) in al_decode_frame() argument
768 out_samples[i], i, q->coding_mode); in al_decode_frame()
778 float *p1 = out_samples[i]; in al_decode_frame()
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| D | g726.c | 474 int out_samples, ret; in g726_decode_frame() local
476 out_samples = buf_size * 8 / c->code_size; in g726_decode_frame()
479 frame->nb_samples = out_samples; in g726_decode_frame()
486 while (out_samples--) in g726_decode_frame()
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| D | imc.c | 104 float *out_samples; member
732 float *dst1 = q->out_samples; in imc_imdct256()
733 float *dst2 = q->out_samples + (COEFFS - 1); in imc_imdct256()
1047 q->out_samples = (float *)frame->extended_data[i]; in imc_decode_frame()
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| D | mpegaudiodec_template.c | 1823 OUT_INT **out_samples; in decode_frame_mp3on4() local
1831 out_samples = (OUT_INT **)frame->extended_data; in decode_frame_mp3on4()
1866 outptr[0] = out_samples[s->coff[fr]]; in decode_frame_mp3on4()
1868 outptr[1] = out_samples[s->coff[fr] + 1]; in decode_frame_mp3on4()
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| D | wavpackenc.c | 545 static void decorr_mono(int32_t *in_samples, int32_t *out_samples, in decorr_mono() argument
553 out_samples += (nb_samples - 1); in decorr_mono()
574 out_samples[0] = left; in decorr_mono()
576 out_samples += dir; in decorr_mono()
590 out_samples[0] = left; in decorr_mono()
592 out_samples += dir; in decorr_mono()
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| /third_party/gstreamer/gstplugins_bad/ext/openmpt/ |
| D | gstopenmptdec.c | 871 int16_t *out_samples = (int16_t *) (map.data); in gst_openmpt_dec_decode() local
877 out_samples); in gst_openmpt_dec_decode()
883 out_samples); in gst_openmpt_dec_decode()
889 out_samples); in gst_openmpt_dec_decode()
898 float *out_samples = (float *) (map.data); in gst_openmpt_dec_decode() local
904 out_samples); in gst_openmpt_dec_decode()
910 out_samples); in gst_openmpt_dec_decode()
916 out_samples); in gst_openmpt_dec_decode()
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| /third_party/ffmpeg/libavfilter/ |
| D | af_axcorrelate.c | 222 const int out_samples = available - s->size; in activate() local
247 out = ff_get_audio_buffer(ctx->outputs[0], out_samples); in activate()
254 s->pts += out_samples; in activate()
256 av_audio_fifo_drain(s->fifo[0], out_samples); in activate()
257 av_audio_fifo_drain(s->fifo[1], out_samples); in activate()
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| D | af_afftdn.c | 64 double *out_samples; member
726 dnch->out_samples = av_calloc(s->buffer_length, sizeof(*dnch->out_samples)); in config_input()
744 !dnch->out_samples || in config_input()
805 dnch->out_samples[i] = 0; in config_input()
1097 double *dst = dnch->out_samples; in filter_channel()
1232 double *src = dnch->out_samples; in output_frame()
1328 av_freep(&dnch->out_samples); in uninit()
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| /third_party/gstreamer/gstplugins_bad/ext/lv2/ |
| D | gstlv2filter.c | 384 guint j, k, l, nframes, samples, out_samples; in gst_lv2_filter_transform_data() local
414 out_samples = nframes / lv2_group->ports->len; in gst_lv2_filter_transform_data()
416 GST_LOG_OBJECT (self, "out: samples=%u, nframes=%u, ports=%d", out_samples, in gst_lv2_filter_transform_data()
421 out + (j * out_samples)); in gst_lv2_filter_transform_data()
444 (gfloat *) out_map->data, out_samples, out); in gst_lv2_filter_transform_data()
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| /third_party/ffmpeg/libswresample/ |
| D | soxr_resample.c | 117 double out_samples = (double)s->out_sample_rate / s->in_sample_rate * in_samples; in get_out_samples() local
123 return (int64_t)(out_samples + delayed_samples + 1 + .5); in get_out_samples()
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| D | swresample.c | 877 int64_t out_samples; in swr_get_out_samples() local
885 out_samples = s->resampler->get_out_samples(s, in_samples); in swr_get_out_samples()
887 out_samples = s->in_buffer_count + in_samples; in swr_get_out_samples()
891 if (out_samples > INT_MAX) in swr_get_out_samples()
894 return out_samples; in swr_get_out_samples()
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| /third_party/gstreamer/gstplugins_bad/gst/speed/ |
| D | gstspeed.c | 588 guint c, in_samples, out_samples, out_size; in speed_chain() local
607 out_samples = 0; in speed_chain()
611 out_samples = speed_chain_int16 (filter, in_buf, out_buf, c, in_samples); in speed_chain()
613 out_samples = in speed_chain()
617 size = out_samples * GST_AUDIO_INFO_BPF (&filter->info); in speed_chain()
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| /third_party/gstreamer/gstplugins_good/ext/pulse/ |
| D | pulsesink.c | 205 guint64 * sample, guchar * data, gint in_samples, gint out_samples,
1392 out_samples -= (d - db)/bpf; \
1410 out_samples -= (d - db)/bpf; \
1427 out_samples -= (d - db)/bpf; \
1444 out_samples -= (d - db)/bpf; \
1452 guchar * data, gint in_samples, gint out_samples, gint * accum) in gst_pulseringbuffer_commit() argument
1495 reverse = out_samples < 0; in gst_pulseringbuffer_commit()
1496 out_samples = ABS (out_samples); in gst_pulseringbuffer_commit()
1498 if (in_samples >= out_samples) in gst_pulseringbuffer_commit()
1501 toprocess = &out_samples; in gst_pulseringbuffer_commit()
[all …]
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| /third_party/mindspore/mindspore/ccsrc/minddata/dataset/engine/gnn/ |
| D | graph_data_impl.h | 266 std::vector<NodeIdType> *out_samples);
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| D | graph_data_impl.cc | 292 int32_t samples_num, std::vector<NodeIdType> *out_samples) { in NegativeSample() argument
301 out_samples->emplace_back(data[shuffled_ids[i]]); in NegativeSample()
302 if (out_samples->size() >= samples_num) { in NegativeSample()
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| /third_party/gstreamer/gstplugins_base/ |
| D | ChangeLog | 16065 out_samples is set and used in the 'no_align' block.
|