mirror of https://git.ffmpeg.org/ffmpeg.git
230 lines
7.8 KiB
C
230 lines
7.8 KiB
C
/*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "common.h"
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#include "samplefmt.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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typedef struct SampleFmtInfo {
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const char *name;
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int bits;
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int planar;
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enum AVSampleFormat altform; ///< planar<->packed alternative form
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} SampleFmtInfo;
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/** this table gives more information about formats */
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static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = {
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[AV_SAMPLE_FMT_U8] = { .name = "u8", .bits = 8, .planar = 0, .altform = AV_SAMPLE_FMT_U8P },
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[AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16, .planar = 0, .altform = AV_SAMPLE_FMT_S16P },
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[AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_S32P },
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[AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_FLTP },
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[AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_DBLP },
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[AV_SAMPLE_FMT_U8P] = { .name = "u8p", .bits = 8, .planar = 1, .altform = AV_SAMPLE_FMT_U8 },
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[AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1, .altform = AV_SAMPLE_FMT_S16 },
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[AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_S32 },
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[AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_FLT },
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[AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_DBL },
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};
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const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return NULL;
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return sample_fmt_info[sample_fmt].name;
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}
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enum AVSampleFormat av_get_sample_fmt(const char *name)
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{
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int i;
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for (i = 0; i < AV_SAMPLE_FMT_NB; i++)
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if (!strcmp(sample_fmt_info[i].name, name))
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return i;
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return AV_SAMPLE_FMT_NONE;
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}
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enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return AV_SAMPLE_FMT_NONE;
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if (sample_fmt_info[sample_fmt].planar)
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return sample_fmt_info[sample_fmt].altform;
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return sample_fmt;
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}
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enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return AV_SAMPLE_FMT_NONE;
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if (sample_fmt_info[sample_fmt].planar)
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return sample_fmt;
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return sample_fmt_info[sample_fmt].altform;
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}
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char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)
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{
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/* print header */
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if (sample_fmt < 0)
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snprintf(buf, buf_size, "name " " depth");
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else if (sample_fmt < AV_SAMPLE_FMT_NB) {
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SampleFmtInfo info = sample_fmt_info[sample_fmt];
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snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
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}
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return buf;
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}
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int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
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{
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return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
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0 : sample_fmt_info[sample_fmt].bits >> 3;
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}
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#if FF_API_GET_BITS_PER_SAMPLE_FMT
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int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt)
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{
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return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
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0 : sample_fmt_info[sample_fmt].bits;
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}
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#endif
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int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
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{
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if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
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return 0;
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return sample_fmt_info[sample_fmt].planar;
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}
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int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
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enum AVSampleFormat sample_fmt, int align)
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{
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int line_size;
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int sample_size = av_get_bytes_per_sample(sample_fmt);
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int planar = av_sample_fmt_is_planar(sample_fmt);
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/* validate parameter ranges */
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if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
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return AVERROR(EINVAL);
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/* auto-select alignment if not specified */
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if (!align) {
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align = 1;
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nb_samples = FFALIGN(nb_samples, 32);
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}
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/* check for integer overflow */
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if (nb_channels > INT_MAX / align ||
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(int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
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return AVERROR(EINVAL);
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line_size = planar ? FFALIGN(nb_samples * sample_size, align) :
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FFALIGN(nb_samples * sample_size * nb_channels, align);
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if (linesize)
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*linesize = line_size;
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return planar ? line_size * nb_channels : line_size;
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}
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int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
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const uint8_t *buf, int nb_channels, int nb_samples,
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enum AVSampleFormat sample_fmt, int align)
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{
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int ch, planar, buf_size, line_size;
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planar = av_sample_fmt_is_planar(sample_fmt);
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buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,
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sample_fmt, align);
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if (buf_size < 0)
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return buf_size;
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audio_data[0] = buf;
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for (ch = 1; planar && ch < nb_channels; ch++)
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audio_data[ch] = audio_data[ch-1] + line_size;
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if (linesize)
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*linesize = line_size;
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return 0;
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}
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int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
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int nb_samples, enum AVSampleFormat sample_fmt, int align)
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{
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uint8_t *buf;
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int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
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sample_fmt, align);
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if (size < 0)
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return size;
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buf = av_malloc(size);
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if (!buf)
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return AVERROR(ENOMEM);
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size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
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nb_samples, sample_fmt, align);
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if (size < 0) {
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av_free(buf);
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return size;
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}
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av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt);
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return 0;
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}
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int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
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int src_offset, int nb_samples, int nb_channels,
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enum AVSampleFormat sample_fmt)
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{
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int planar = av_sample_fmt_is_planar(sample_fmt);
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int planes = planar ? nb_channels : 1;
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int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
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int data_size = nb_samples * block_align;
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int i;
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dst_offset *= block_align;
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src_offset *= block_align;
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for (i = 0; i < planes; i++)
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memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
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return 0;
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}
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int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
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int nb_channels, enum AVSampleFormat sample_fmt)
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{
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int planar = av_sample_fmt_is_planar(sample_fmt);
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int planes = planar ? nb_channels : 1;
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int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
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int data_size = nb_samples * block_align;
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int fill_char = (sample_fmt == AV_SAMPLE_FMT_U8 ||
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sample_fmt == AV_SAMPLE_FMT_U8P) ? 0x80 : 0x00;
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int i;
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offset *= block_align;
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for (i = 0; i < planes; i++)
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memset(audio_data[i] + offset, fill_char, data_size);
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return 0;
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}
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