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Searched refs:nb_samples (Results 1 - 25 of 307) sorted by relevance

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/third_party/ffmpeg/libavutil/
H A Daudio_fifo.c40 int nb_samples; /**< number of samples currently in the FIFO */ member
63 int nb_samples) in av_audio_fifo_alloc()
69 if (av_samples_get_buffer_size(&buf_size, channels, nb_samples, sample_fmt, 1) < 0) in av_audio_fifo_alloc()
78 af->sample_size = buf_size / nb_samples; in av_audio_fifo_alloc()
90 af->allocated_samples = nb_samples; in av_audio_fifo_alloc()
99 int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples) in av_audio_fifo_realloc() argument
105 if ((ret = av_samples_get_buffer_size(&buf_size, af->channels, nb_samples, in av_audio_fifo_realloc()
115 af->allocated_samples = nb_samples; in av_audio_fifo_realloc()
119 int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples) in av_audio_fifo_write() argument
124 if (av_audio_fifo_space(af) < nb_samples) { in av_audio_fifo_write()
62 av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples) av_audio_fifo_alloc() argument
145 av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples) av_audio_fifo_peek() argument
150 av_audio_fifo_peek_at(AVAudioFifo *af, void **data, int nb_samples, int offset) av_audio_fifo_peek_at() argument
174 av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples) av_audio_fifo_read() argument
194 av_audio_fifo_drain(AVAudioFifo *af, int nb_samples) av_audio_fifo_drain() argument
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H A Dsamplefmt.c121 int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, in av_samples_get_buffer_size() argument
129 if (!sample_size || nb_samples <= 0 || nb_channels <= 0) in av_samples_get_buffer_size()
134 if (nb_samples > INT_MAX - 31) in av_samples_get_buffer_size()
137 nb_samples = FFALIGN(nb_samples, 32); in av_samples_get_buffer_size()
142 (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size) in av_samples_get_buffer_size()
145 line_size = planar ? FFALIGN(nb_samples * sample_size, align) : in av_samples_get_buffer_size()
146 FFALIGN(nb_samples * sample_size * nb_channels, align); in av_samples_get_buffer_size()
154 const uint8_t *buf, int nb_channels, int nb_samples, in av_samples_fill_arrays()
160 buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples, in av_samples_fill_arrays()
153 av_samples_fill_arrays(uint8_t **audio_data, int *linesize, const uint8_t *buf, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align) av_samples_fill_arrays() argument
182 av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align) av_samples_alloc() argument
207 av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align) av_samples_alloc_array_and_samples() argument
222 av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset, int src_offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt) av_samples_copy() argument
246 av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt) av_samples_set_silence() argument
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H A Daudio_fifo.h62 * @param nb_samples initial allocation size, in samples
66 int nb_samples);
72 * @param nb_samples new allocation size, in samples
76 int av_audio_fifo_realloc(AVAudioFifo *af, int nb_samples);
82 * is less than nb_samples.
89 * @param nb_samples number of samples to write
92 * actually written will always be nb_samples.
94 int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples);
104 * @param nb_samples number of samples to peek
107 * be greater than nb_samples, an
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/third_party/ffmpeg/libavresample/
H A Daudio_data.c74 int channels, int nb_samples, in ff_audio_data_init()
104 a->allocated_samples = nb_samples * !read_only; in ff_audio_data_init()
105 a->nb_samples = nb_samples; in ff_audio_data_init()
119 AudioData *ff_audio_data_alloc(int channels, int nb_samples, in ff_audio_data_alloc() argument
149 if (nb_samples > 0) { in ff_audio_data_alloc()
150 ret = ff_audio_data_realloc(a, nb_samples); in ff_audio_data_alloc()
162 int ff_audio_data_realloc(AudioData *a, int nb_samples) in ff_audio_data_realloc() argument
167 if (a->allocated_samples >= nb_samples) in ff_audio_data_realloc()
175 a->allocated_channels, nb_samples, in ff_audio_data_realloc()
73 ff_audio_data_init(AudioData *a, uint8_t * const *src, int plane_size, int channels, int nb_samples, enum AVSampleFormat sample_fmt, int read_only, const char *name) ff_audio_data_init() argument
278 ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, int src_offset, int nb_samples) ff_audio_data_combine() argument
334 ff_audio_data_drain(AudioData *a, int nb_samples) ff_audio_data_drain() argument
351 ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, int nb_samples) ff_audio_data_add_to_fifo() argument
366 ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples) ff_audio_data_read_from_fifo() argument
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H A Daudio_data.h43 int nb_samples; /**< current number of samples */ member
73 * @param nb_samples number of samples in the source data
80 int channels, int nb_samples,
90 * @param nb_samples number of samples to allocate space for
95 AudioData *ff_audio_data_alloc(int channels, int nb_samples,
105 * @param nb_samples number of samples to allocate space for
108 int ff_audio_data_realloc(AudioData *a, int nb_samples);
138 * @param nb_samples number of samples to copy
142 int src_offset, int nb_samples);
150 * @param nb_samples numbe
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/third_party/ffmpeg/libavfilter/
H A Daf_afade.c38 int64_t nb_samples; member
48 int nb_samples, int channels, int direction,
52 int nb_samples, int channels,
150 int nb_samples, int channels, int dir, \
155 for (i = 0; i < nb_samples; i++) { \
168 int nb_samples, int channels, int dir, \
175 for (i = 0; i < nb_samples; i++) { \
209 s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE); in config_output()
227 { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT6
274 int nb_samples = buf->nb_samples; filter_frame() local
451 int ret = 0, nb_samples, status; activate() local
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H A Df_loop.c45 int64_t nb_samples; member
93 static int push_samples(AVFilterContext *ctx, int nb_samples) in push_samples() argument
100 while (s->loop != 0 && i < nb_samples) { in push_samples()
101 out = ff_get_audio_buffer(outlink, FFMIN(nb_samples, s->nb_samples - s->current_sample)); in push_samples()
104 ret = av_audio_fifo_peek_at(s->fifo, (void **)out->extended_data, out->nb_samples, s->current_sample); in push_samples()
110 out->nb_samples = ret; in push_samples()
111 s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base); in push_samples()
112 i += out->nb_samples; in push_samples()
113 s->current_sample += out->nb_samples; in push_samples()
166 int nb_samples = frame->nb_samples; afilter_frame() local
190 int nb_samples = av_audio_fifo_size(s->left); arequest_frame() local
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H A Daudio.c32 AVFrame *ff_null_get_audio_buffer(AVFilterLink *link, int nb_samples) in ff_null_get_audio_buffer() argument
34 return ff_get_audio_buffer(link->dst->outputs[0], nb_samples); in ff_null_get_audio_buffer()
37 AVFrame *ff_default_get_audio_buffer(AVFilterLink *link, int nb_samples) in ff_default_get_audio_buffer() argument
52 nb_samples, link->format, align); in ff_default_get_audio_buffer()
67 if (pool_channels != channels || pool_nb_samples < nb_samples || in ff_default_get_audio_buffer()
72 nb_samples, link->format, align); in ff_default_get_audio_buffer()
82 frame->nb_samples = nb_samples; in ff_default_get_audio_buffer()
95 av_samples_set_silence(frame->extended_data, 0, nb_samples, channels, link->format); in ff_default_get_audio_buffer()
100 AVFrame *ff_get_audio_buffer(AVFilterLink *link, int nb_samples) in ff_get_audio_buffer() argument
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H A Daf_amix.c58 int nb_samples; member
73 int nb_samples; member
87 frame_list->nb_samples = 0; in frame_list_clear()
96 return frame_list->list->nb_samples; in frame_list_next_frame_size()
106 static void frame_list_remove_samples(FrameList *frame_list, int nb_samples) in frame_list_remove_samples() argument
108 if (nb_samples >= frame_list->nb_samples) { in frame_list_remove_samples()
111 int samples = nb_samples; in frame_list_remove_samples()
115 if (info->nb_samples <= samples) { in frame_list_remove_samples()
116 samples -= info->nb_samples; in frame_list_remove_samples()
133 frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts) frame_list_add_frame() argument
213 calculate_scales(MixContext *s, int nb_samples) calculate_scales() argument
302 int nb_samples, ns, i; output_frame() local
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H A Daf_adenorm.c43 const void *src, int nb_samples);
47 const void *srcp, int nb_samples) in dc_denorm_fltp()
54 for (int n = 0; n < nb_samples; n++) { in dc_denorm_fltp()
60 const void *srcp, int nb_samples) in dc_denorm_dblp()
67 for (int n = 0; n < nb_samples; n++) { in dc_denorm_dblp()
73 const void *srcp, int nb_samples) in ac_denorm_fltp()
81 for (int n = 0; n < nb_samples; n++) { in ac_denorm_fltp()
87 const void *srcp, int nb_samples) in ac_denorm_dblp()
95 for (int n = 0; n < nb_samples; n++) { in ac_denorm_dblp()
101 const void *srcp, int nb_samples) in sq_denorm_fltp()
46 dc_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) dc_denorm_fltp() argument
59 dc_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) dc_denorm_dblp() argument
72 ac_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) ac_denorm_fltp() argument
86 ac_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) ac_denorm_dblp() argument
100 sq_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) sq_denorm_fltp() argument
114 sq_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) sq_denorm_dblp() argument
128 ps_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) ps_denorm_fltp() argument
142 ps_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples) ps_denorm_dblp() argument
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H A Daf_rubberband.c41 int nb_samples; member
98 int ret = 0, nb_samples; in filter_frame() local
103 rubberband_process(s->rbs, (const float *const *)in->data, in->nb_samples, ff_outlink_get_status(inlink)); in filter_frame()
104 s->nb_samples_in += in->nb_samples; in filter_frame()
106 nb_samples = rubberband_available(s->rbs); in filter_frame()
107 if (nb_samples > 0) { in filter_frame()
108 out = ff_get_audio_buffer(outlink, nb_samples); in filter_frame()
116 nb_samples = rubberband_retrieve(s->rbs, (float *const *)out->data, nb_samples); in filter_frame()
117 out->nb_samples in filter_frame()
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H A Dtrim.c58 int64_t nb_samples; member
239 s->next_pts = pts + frame->nb_samples; in atrim_filter_frame()
246 start_sample = frame->nb_samples; in atrim_filter_frame()
249 s->nb_samples + frame->nb_samples > s->start_sample) { in atrim_filter_frame()
251 start_sample = FFMIN(start_sample, s->start_sample - s->nb_samples); in atrim_filter_frame()
255 pts + frame->nb_samples > s->start_pts) { in atrim_filter_frame()
269 end_sample = frame->nb_samples; in atrim_filter_frame()
275 s->nb_samples < s->end_sample) { in atrim_filter_frame()
277 end_sample = FFMAX(end_sample, s->end_sample - s->nb_samples); in atrim_filter_frame()
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H A Daf_volumedetect.c40 int nb_samples = samples->nb_samples; in filter_frame() local
47 nb_samples *= nb_channels; in filter_frame()
52 for (i = 0; i < nb_samples; i++) in filter_frame()
73 uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0; in print_stats() local
77 nb_samples += vd->histogram[i]; in print_stats()
78 av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples); in print_stats()
79 if (!nb_samples) in print_stats()
82 /* If nb_samples > 1<<34, there is a risk of overflow in the in print_stats()
86 shift = av_log2(nb_samples >> 3 in print_stats()
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H A Daf_acontrast.c32 int nb_samples, int channels, float contrast);
46 int nb_samples, int channels, in filter_flt()
53 for (n = 0; n < nb_samples; n++) { in filter_flt()
66 int nb_samples, int channels, in filter_dbl()
73 for (n = 0; n < nb_samples; n++) { in filter_dbl()
86 int nb_samples, int channels, in filter_fltp()
95 for (n = 0; n < nb_samples; n++) { in filter_fltp()
104 int nb_samples, int channels, in filter_dblp()
113 for (n = 0; n < nb_samples; n++) { in filter_dblp()
146 out = ff_get_audio_buffer(outlink, in->nb_samples); in filter_frame()
45 filter_flt(void **d, const void **s, int nb_samples, int channels, float contrast) filter_flt() argument
65 filter_dbl(void **d, const void **s, int nb_samples, int channels, float contrast) filter_dbl() argument
85 filter_fltp(void **d, const void **s, int nb_samples, int channels, float contrast) filter_fltp() argument
103 filter_dblp(void **d, const void **s, int nb_samples, int channels, float contrast) filter_dblp() argument
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H A Df_reverse.c38 int64_t nb_samples; member
157 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_planar()
163 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_planar()
169 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_planar()
175 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_planar()
181 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_planar()
187 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_planar()
202 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_packed()
209 for (int i = 0, j = out->nb_samples - 1; i < j; i++, j--) in reverse_samples_packed()
216 for (int i = 0, j = out->nb_samples in reverse_samples_packed()
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H A Dasrc_anoisesrc.c37 int nb_samples; member
77 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
78 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
194 int nb_samples, i; in activate() local
203 } else if (!s->infinite && s->duration < s->nb_samples) { in activate()
204 nb_samples = s->duration; in activate()
206 nb_samples = s->nb_samples; in activate()
209 if (!(frame = ff_get_audio_buffer(outlink, nb_samples))) in activate()
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H A Dasrc_hilbert.c34 int nb_samples; member
49 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
50 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
132 int nb_samples; in activate() local
137 nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts); in activate()
138 if (nb_samples <= 0) { in activate()
143 if (!(frame = ff_get_audio_buffer(outlink, nb_samples))) in activate()
146 memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeo in activate()
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H A Daf_amultiply.c47 int nb_samples; in activate() local
52 nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), in activate()
54 for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) { in activate()
58 if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) { in activate()
59 ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frames[i]); in activate()
70 plane_samples = FFALIGN(s->frames[0]->nb_samples, s->samples_align); in activate()
72 plane_samples = FFALIGN(s->frames[0]->nb_samples * s->channels, s->samples_align); in activate()
74 out = ff_get_audio_buffer(ctx->outputs[0], s->frames[0]->nb_samples); in activate()
105 if (!nb_samples) { in activate()
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H A Daf_adelay.c50 void (*delay_channel)(ChanDelay *d, int nb_samples,
67 static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
74 while (nb_samples) { \
76 const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
83 nb_samples -= len; \
87 nb_samples--; \
329 out_frame = ff_get_audio_buffer(outlink, frame->nb_samples); in filter_frame()
342 memcpy(dst, src, frame->nb_samples * s->block_align); in filter_frame()
344 s->delay_channel(d, frame->nb_samples, src, dst); in filter_frame()
348 s->next_pts += av_rescale_q(frame->nb_samples, (AVRationa in filter_frame()
365 int nb_samples = FFMIN(s->padding, 2048); activate() local
397 int nb_samples = FFMIN(s->max_delay, 2048); activate() local
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/third_party/ffmpeg/libavcodec/
H A Dalac.c82 int nb_samples; /**< number of samples in the current frame */ member
113 int nb_samples, int bps, int rice_history_mult) in rice_decompress()
119 for (i = 0; i < nb_samples; i++) { in rice_decompress()
142 if ((history < 128) && (i + 1 < nb_samples)) { in rice_decompress()
151 if (block_size >= nb_samples - i) { in rice_decompress()
154 nb_samples, i); in rice_decompress()
155 block_size = nb_samples - i - 1; in rice_decompress()
175 int nb_samples, int bps, int16_t *lpc_coefs, in lpc_prediction()
184 if (nb_samples <= 1) in lpc_prediction()
189 (nb_samples in lpc_prediction()
112 rice_decompress(ALACContext *alac, int32_t *output_buffer, int nb_samples, int bps, int rice_history_mult) rice_decompress() argument
174 lpc_prediction(int32_t *error_buffer, uint32_t *buffer_out, int nb_samples, int bps, int16_t *lpc_coefs, int lpc_order, int lpc_quant) lpc_prediction() argument
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H A Daudio_frame_queue.c53 new->duration = f->nb_samples; in ff_af_queue_add()
68 afq->remaining_samples += f->nb_samples; in ff_af_queue_add()
75 void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, in ff_af_queue_remove() argument
87 av_log(afq->avctx, AV_LOG_WARNING, "Trying to remove %d samples, but the queue is empty\n", nb_samples); in ff_af_queue_remove()
91 for(i=0; nb_samples && i<afq->frame_count; i++){ in ff_af_queue_remove()
92 int n= FFMIN(afq->frames[i].duration, nb_samples); in ff_af_queue_remove()
94 nb_samples -= n; in ff_af_queue_remove()
104 if(nb_samples){ in ff_af_queue_remove()
108 afq->frames[0].pts += nb_samples; in ff_af_queue_remove()
109 av_log(afq->avctx, AV_LOG_DEBUG, "Trying to remove %d more samples than there are in the queue\n", nb_samples); in ff_af_queue_remove()
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H A Dlibopusdec.c163 int ret, nb_samples; in libopus_decode() local
165 frame->nb_samples = MAX_FRAME_SIZE; in libopus_decode()
170 nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size, in libopus_decode()
172 frame->nb_samples, 0); in libopus_decode()
174 nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size, in libopus_decode()
176 frame->nb_samples, 0); in libopus_decode()
178 if (nb_samples < 0) { in libopus_decode()
180 opus_strerror(nb_samples)); in libopus_decode()
181 return ff_opus_error_to_averror(nb_samples); in libopus_decode()
186 int i = avc->ch_layout.nb_channels * nb_samples; in libopus_decode()
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H A Dadpcm.c850 int nb_samples = 0; in get_nb_samples() local
866 nb_samples = 128; in get_nb_samples()
871 nb_samples = 64; in get_nb_samples()
886 nb_samples = buf_size * 2 / ch; in get_nb_samples()
889 if (nb_samples) in get_nb_samples()
890 return nb_samples; in get_nb_samples()
912 nb_samples = FFMIN((buf_size - 8) * 2, *coded_samples); in get_nb_samples()
919 nb_samples = (buf_size - 12) / 30 * 28; in get_nb_samples()
924 nb_samples = (buf_size - (4 + 8 * ch)) * 2 / ch; in get_nb_samples()
927 nb_samples in get_nb_samples()
1075 int nb_samples, coded_samples, approx_nb_samples, ret; adpcm_decode_frame() local
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/third_party/ffmpeg/libavutil/tests/
H A Daudio_fifo.c64 static void print_audio_bytes(const TestStruct *test_sample, void **data_planes, int nb_samples) in print_audio_bytes() argument
70 int line_size = (buffers > 1) ? nb_samples * byte_offset in print_audio_bytes()
71 : nb_samples * byte_offset * test_sample->nb_ch; in print_audio_bytes()
84 static int read_samples_from_audio_fifo(AVAudioFifo* afifo, void ***output, int nb_samples) in read_samples_from_audio_fifo() argument
87 int samples = FFMIN(nb_samples, afifo->nb_samples); in read_samples_from_audio_fifo()
103 return av_audio_fifo_read(afifo, *output, nb_samples); in read_samples_from_audio_fifo()
107 int nb_samples, int offset) in write_samples_to_audio_fifo()
112 if(nb_samples > test_sample->nb_samples_pch - offset){ in write_samples_to_audio_fifo()
124 return av_audio_fifo_write(afifo, data_planes, nb_samples); in write_samples_to_audio_fifo()
106 write_samples_to_audio_fifo(AVAudioFifo* afifo, const TestStruct *test_sample, int nb_samples, int offset) write_samples_to_audio_fifo() argument
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/third_party/ffmpeg/libavformat/
H A Duncodedframecrcenc.c80 int nb_planes, nb_samples, p; in audio_frame_cksum() local
84 nb_samples = frame->nb_samples; in audio_frame_cksum()
86 nb_samples *= nb_planes; in audio_frame_cksum()
90 av_bprintf(bp, ", %d samples", frame->nb_samples); in audio_frame_cksum()
98 cksum_line_u8(&cksum, d, nb_samples); in audio_frame_cksum()
102 cksum_line_s16(&cksum, d, nb_samples); in audio_frame_cksum()
106 cksum_line_s32(&cksum, d, nb_samples); in audio_frame_cksum()
110 cksum_line_flt(&cksum, d, nb_samples); in audio_frame_cksum()
114 cksum_line_dbl(&cksum, d, nb_samples); in audio_frame_cksum()
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