mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-22 15:23:11 +00:00
61930bd0d7
* qatar/master: (27 commits) libxvid: Give more suitable names to libxvid-related files. libxvid: Separate libxvid encoder from libxvid rate control code. jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse(). fate: cosmetics: lowercase some comments fate: Give more consistent names to some RealVideo/RealAudio tests. lavfi: add avfilter_get_audio_buffer_ref_from_arrays(). lavfi: add extended_data to AVFilterBuffer. lavc: check that extended_data is properly set in avcodec_encode_audio2(). lavc: pad last audio frame with silence when needed. samplefmt: add a function for filling a buffer with silence. samplefmt: add a function for copying audio samples. lavr: do not try to copy to uninitialized output audio data. lavr: make avresample_read() with NULL output discard samples. fate: split idroq audio and video into separate tests fate: improve dependencies fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests fate: split some combined tests into separate audio and video tests fate: fix dependencies for probe tests mips: intreadwrite: fix inline asm for gcc 4.8 mips: intreadwrite: remove unnecessary inline asm ... Conflicts: cmdutils.h configure doc/APIchanges doc/filters.texi ffmpeg.c ffplay.c libavcodec/internal.h libavcodec/jpeglsdec.c libavcodec/libschroedingerdec.c libavcodec/libxvid.c libavcodec/libxvid_rc.c libavcodec/utils.c libavcodec/version.h libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/buffersink.h tests/Makefile tests/fate/aac.mak tests/fate/audio.mak tests/fate/demux.mak tests/fate/ea.mak tests/fate/image.mak tests/fate/libavutil.mak tests/fate/lossless-audio.mak tests/fate/lossless-video.mak tests/fate/microsoft.mak tests/fate/qt.mak tests/fate/real.mak tests/fate/screen.mak tests/fate/video.mak tests/fate/voice.mak tests/fate/vqf.mak tests/ref/fate/ea-mad tests/ref/fate/ea-tqi Merged-by: Michael Niedermayer <michaelni@gmx.at>
235 lines
8.0 KiB
C
235 lines
8.0 KiB
C
/*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
#include "samplefmt.h"
|
|
|
|
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
|
|
typedef struct SampleFmtInfo {
|
|
char name[8];
|
|
int bits;
|
|
int planar;
|
|
enum AVSampleFormat altform; ///< planar<->packed alternative form
|
|
} SampleFmtInfo;
|
|
|
|
/** this table gives more information about formats */
|
|
static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = {
|
|
[AV_SAMPLE_FMT_U8] = { .name = "u8", .bits = 8, .planar = 0, .altform = AV_SAMPLE_FMT_U8P },
|
|
[AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16, .planar = 0, .altform = AV_SAMPLE_FMT_S16P },
|
|
[AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_S32P },
|
|
[AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_FLTP },
|
|
[AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_DBLP },
|
|
[AV_SAMPLE_FMT_U8P] = { .name = "u8p", .bits = 8, .planar = 1, .altform = AV_SAMPLE_FMT_U8 },
|
|
[AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1, .altform = AV_SAMPLE_FMT_S16 },
|
|
[AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_S32 },
|
|
[AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_FLT },
|
|
[AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_DBL },
|
|
};
|
|
|
|
const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
|
|
{
|
|
if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
|
|
return NULL;
|
|
return sample_fmt_info[sample_fmt].name;
|
|
}
|
|
|
|
enum AVSampleFormat av_get_sample_fmt(const char *name)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < AV_SAMPLE_FMT_NB; i++)
|
|
if (!strcmp(sample_fmt_info[i].name, name))
|
|
return i;
|
|
return AV_SAMPLE_FMT_NONE;
|
|
}
|
|
|
|
enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar)
|
|
{
|
|
if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
|
|
return AV_SAMPLE_FMT_NONE;
|
|
if (sample_fmt_info[sample_fmt].planar == planar)
|
|
return sample_fmt;
|
|
return sample_fmt_info[sample_fmt].altform;
|
|
}
|
|
|
|
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
|
|
{
|
|
if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
|
|
return AV_SAMPLE_FMT_NONE;
|
|
if (sample_fmt_info[sample_fmt].planar)
|
|
return sample_fmt_info[sample_fmt].altform;
|
|
return sample_fmt;
|
|
}
|
|
|
|
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
|
|
{
|
|
if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
|
|
return AV_SAMPLE_FMT_NONE;
|
|
if (sample_fmt_info[sample_fmt].planar)
|
|
return sample_fmt;
|
|
return sample_fmt_info[sample_fmt].altform;
|
|
}
|
|
|
|
char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)
|
|
{
|
|
/* print header */
|
|
if (sample_fmt < 0)
|
|
snprintf(buf, buf_size, "name " " depth");
|
|
else if (sample_fmt < AV_SAMPLE_FMT_NB) {
|
|
SampleFmtInfo info = sample_fmt_info[sample_fmt];
|
|
snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
|
|
}
|
|
|
|
return buf;
|
|
}
|
|
|
|
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
|
|
{
|
|
return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
|
|
0 : sample_fmt_info[sample_fmt].bits >> 3;
|
|
}
|
|
|
|
#if FF_API_GET_BITS_PER_SAMPLE_FMT
|
|
int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt)
|
|
{
|
|
return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
|
|
0 : sample_fmt_info[sample_fmt].bits;
|
|
}
|
|
#endif
|
|
|
|
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
|
|
{
|
|
if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
|
|
return 0;
|
|
return sample_fmt_info[sample_fmt].planar;
|
|
}
|
|
|
|
int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
|
|
enum AVSampleFormat sample_fmt, int align)
|
|
{
|
|
int line_size;
|
|
int sample_size = av_get_bytes_per_sample(sample_fmt);
|
|
int planar = av_sample_fmt_is_planar(sample_fmt);
|
|
|
|
/* validate parameter ranges */
|
|
if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
|
|
return AVERROR(EINVAL);
|
|
|
|
/* auto-select alignment if not specified */
|
|
if (!align) {
|
|
align = 1;
|
|
nb_samples = FFALIGN(nb_samples, 32);
|
|
}
|
|
|
|
/* check for integer overflow */
|
|
if (nb_channels > INT_MAX / align ||
|
|
(int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
|
|
return AVERROR(EINVAL);
|
|
|
|
line_size = planar ? FFALIGN(nb_samples * sample_size, align) :
|
|
FFALIGN(nb_samples * sample_size * nb_channels, align);
|
|
if (linesize)
|
|
*linesize = line_size;
|
|
|
|
return planar ? line_size * nb_channels : line_size;
|
|
}
|
|
|
|
int 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)
|
|
{
|
|
int ch, planar, buf_size, line_size;
|
|
|
|
planar = av_sample_fmt_is_planar(sample_fmt);
|
|
buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,
|
|
sample_fmt, align);
|
|
if (buf_size < 0)
|
|
return buf_size;
|
|
|
|
audio_data[0] = buf;
|
|
for (ch = 1; planar && ch < nb_channels; ch++)
|
|
audio_data[ch] = audio_data[ch-1] + line_size;
|
|
|
|
if (linesize)
|
|
*linesize = line_size;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
|
|
int nb_samples, enum AVSampleFormat sample_fmt, int align)
|
|
{
|
|
uint8_t *buf;
|
|
int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
|
|
sample_fmt, align);
|
|
if (size < 0)
|
|
return size;
|
|
|
|
buf = av_mallocz(size);
|
|
if (!buf)
|
|
return AVERROR(ENOMEM);
|
|
|
|
size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
|
|
nb_samples, sample_fmt, align);
|
|
if (size < 0) {
|
|
av_free(buf);
|
|
return size;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int 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)
|
|
{
|
|
int planar = av_sample_fmt_is_planar(sample_fmt);
|
|
int planes = planar ? nb_channels : 1;
|
|
int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
|
|
int data_size = nb_samples * block_align;
|
|
int i;
|
|
|
|
dst_offset *= block_align;
|
|
src_offset *= block_align;
|
|
|
|
for (i = 0; i < planes; i++)
|
|
memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
|
|
int nb_channels, enum AVSampleFormat sample_fmt)
|
|
{
|
|
int planar = av_sample_fmt_is_planar(sample_fmt);
|
|
int planes = planar ? nb_channels : 1;
|
|
int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
|
|
int data_size = nb_samples * block_align;
|
|
int fill_char = (sample_fmt == AV_SAMPLE_FMT_U8 ||
|
|
sample_fmt == AV_SAMPLE_FMT_U8P) ? 0x80 : 0x00;
|
|
int i;
|
|
|
|
offset *= block_align;
|
|
|
|
for (i = 0; i < planes; i++)
|
|
memset(audio_data[i] + offset, fill_char, data_size);
|
|
|
|
return 0;
|
|
}
|