ffmpeg/libavutil/samplefmt.c
Michael Niedermayer 61930bd0d7 Merge remote-tracking branch 'qatar/master'
* 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>
2012-05-10 02:25:41 +02:00

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;
}