libavutil/samplefmt.c - third_party/ffmpeg - Git at Google

 /*
  * 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 "common.h"
 #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_S64]  = { .name =  "s64", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_S64P },
     [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_S64P] = { .name = "s64p", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_S64  },
     [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;
 }

 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) {
         if (nb_samples > INT_MAX - 31)
             return AVERROR(EINVAL);
         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] = (uint8_t *)buf;
     for (ch = 1; planar && ch < nb_channels; ch++)
         audio_data[ch] = audio_data[ch-1] + line_size;

     if (linesize)
         *linesize = line_size;

     return buf_size;
 }

 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_malloc(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;
     }

     av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt);

     return size;
 }

 int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
                                        int nb_samples, enum AVSampleFormat sample_fmt, int align)
 {
     int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;

     *audio_data = av_calloc(nb_planes, sizeof(**audio_data));
     if (!*audio_data)
         return AVERROR(ENOMEM);
     ret = av_samples_alloc(*audio_data, linesize, nb_channels,
                            nb_samples, sample_fmt, align);
     if (ret < 0)
         av_freep(audio_data);
     return ret;
 }

 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;

     if((dst[0] < src[0] ? src[0] - dst[0] : dst[0] - src[0]) >= data_size) {
         for (i = 0; i < planes; i++)
             memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
     } else {
         for (i = 0; i < planes; i++)
             memmove(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;
 }
	/*
	* 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 "common.h"
	#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_S64] = { .name = "s64", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_S64P },
	[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_S64P] = { .name = "s64p", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_S64 },
	[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;
	}

	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) {
	if (nb_samples > INT_MAX - 31)
	return AVERROR(EINVAL);
	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] = (uint8_t *)buf;
	for (ch = 1; planar && ch < nb_channels; ch++)
	audio_data[ch] = audio_data[ch-1] + line_size;

	if (linesize)
	*linesize = line_size;

	return buf_size;
	}

	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_malloc(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;
	}

	av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt);

	return size;
	}

	int av_samples_alloc_array_and_samples(uint8_t **audio_data, int linesize, int nb_channels,
	int nb_samples, enum AVSampleFormat sample_fmt, int align)
	{
	int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;

	audio_data = av_calloc(nb_planes, sizeof(*audio_data));
	if (!*audio_data)
	return AVERROR(ENOMEM);
	ret = av_samples_alloc(*audio_data, linesize, nb_channels,
	nb_samples, sample_fmt, align);
	if (ret < 0)
	av_freep(audio_data);
	return ret;
	}

	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;

	if((dst[0] < src[0] ? src[0] - dst[0] : dst[0] - src[0]) >= data_size) {
	for (i = 0; i < planes; i++)
	memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
	} else {
	for (i = 0; i < planes; i++)
	memmove(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;
	}