1
0
mirror of https://github.com/mpv-player/mpv synced 2024-12-25 08:12:17 +00:00
mpv/audio/format.c
wm4 eec7f61b5f audio/format: change license to LGPL
Although the origins lie somewhere in libaf, which was written by
"anders" and who explicitly disagreed with the LGPL relicensing, we can
change the license of these files, because all code was written by
"alex", who agreed with the relicensing.

The only things that remain from anders' code is the AF_FORMAT_ and af_
prefixes (see e.g. 66f4e563). It was alex who redid this file and added
the format identifiers we have today (507121f7). It's also nice to see
that alex actually claimed copyright on format.c (221a599f). In commit
efb50cab even the bitmask concept (which anders introduced with his
early af_format.c code) was removed, and essentially all lines and
symbols by anders were dropped.

To put it into perspective: the original af_format code was for
converting actual sample data and relied on OSS sample format
identifiers, mpv's format.c/h provides its own sample formats, but
does not do any data conversion.

Remove an now inaccurate comment from format.c (it somehow even survived
the typo that was present in the original commit). Also remove most of
the format.c include statements - most of them are technically anders'
code. We keep limits.h though.
2017-06-20 15:37:28 +02:00

289 lines
9.4 KiB
C

/*
* Copyright (C) 2005 Alex Beregszaszi
*
* This file is part of mpv.
*
* mpv 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.
*
* mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <limits.h>
#include "common/common.h"
#include "audio/filter/af.h"
// number of bytes per sample, 0 if invalid/unknown
int af_fmt_to_bytes(int format)
{
switch (af_fmt_from_planar(format)) {
case AF_FORMAT_U8: return 1;
case AF_FORMAT_S16: return 2;
case AF_FORMAT_S24: return 3;
case AF_FORMAT_S32: return 4;
case AF_FORMAT_FLOAT: return 4;
case AF_FORMAT_DOUBLE: return 8;
}
if (af_fmt_is_spdif(format))
return 2;
return 0;
}
int af_fmt_change_bytes(int format, int bytes)
{
if (!af_fmt_is_valid(format) || !bytes)
return 0;
for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
if (af_fmt_to_bytes(fmt) == bytes &&
af_fmt_is_float(fmt) == af_fmt_is_float(format) &&
af_fmt_is_planar(fmt) == af_fmt_is_planar(format) &&
(fmt == format || (!af_fmt_is_spdif(fmt) && !af_fmt_is_spdif(format))))
return fmt;
}
return 0;
}
// All formats are considered signed, except explicitly unsigned int formats.
bool af_fmt_is_unsigned(int format)
{
return format == AF_FORMAT_U8 || format == AF_FORMAT_U8P;
}
bool af_fmt_is_float(int format)
{
format = af_fmt_from_planar(format);
return format == AF_FORMAT_FLOAT || format == AF_FORMAT_DOUBLE;
}
// true for both unsigned and signed ints
bool af_fmt_is_int(int format)
{
return format && !af_fmt_is_spdif(format) && !af_fmt_is_float(format);
}
// false for interleaved and AF_FORMAT_UNKNOWN
bool af_fmt_is_planar(int format)
{
return format && af_fmt_to_planar(format) == format;
}
bool af_fmt_is_spdif(int format)
{
return af_format_sample_alignment(format) > 1;
}
bool af_fmt_is_pcm(int format)
{
return af_fmt_is_valid(format) && !af_fmt_is_spdif(format);
}
static const int planar_formats[][2] = {
{AF_FORMAT_U8P, AF_FORMAT_U8},
{AF_FORMAT_S16P, AF_FORMAT_S16},
{AF_FORMAT_S32P, AF_FORMAT_S32},
{AF_FORMAT_FLOATP, AF_FORMAT_FLOAT},
{AF_FORMAT_DOUBLEP, AF_FORMAT_DOUBLE},
};
// Return the planar format corresponding to the given format.
// If the format is already planar, return it.
// Return 0 if there's no equivalent.
int af_fmt_to_planar(int format)
{
for (int n = 0; n < MP_ARRAY_SIZE(planar_formats); n++) {
if (planar_formats[n][1] == format)
return planar_formats[n][0];
if (planar_formats[n][0] == format)
return format;
}
return 0;
}
// Return the interleaved format corresponding to the given format.
// If the format is already interleaved, return it.
// Always succeeds if format is actually planar; otherwise return 0.
int af_fmt_from_planar(int format)
{
for (int n = 0; n < MP_ARRAY_SIZE(planar_formats); n++) {
if (planar_formats[n][0] == format)
return planar_formats[n][1];
}
return format;
}
bool af_fmt_is_valid(int format)
{
return format > 0 && format < AF_FORMAT_COUNT;
}
const char *af_fmt_to_str(int format)
{
switch (format) {
case AF_FORMAT_U8: return "u8";
case AF_FORMAT_S16: return "s16";
case AF_FORMAT_S24: return "s24";
case AF_FORMAT_S32: return "s32";
case AF_FORMAT_FLOAT: return "float";
case AF_FORMAT_DOUBLE: return "double";
case AF_FORMAT_U8P: return "u8p";
case AF_FORMAT_S16P: return "s16p";
case AF_FORMAT_S32P: return "s32p";
case AF_FORMAT_FLOATP: return "floatp";
case AF_FORMAT_DOUBLEP: return "doublep";
case AF_FORMAT_S_AAC: return "spdif-aac";
case AF_FORMAT_S_AC3: return "spdif-ac3";
case AF_FORMAT_S_DTS: return "spdif-dts";
case AF_FORMAT_S_DTSHD: return "spdif-dtshd";
case AF_FORMAT_S_EAC3: return "spdif-eac3";
case AF_FORMAT_S_MP3: return "spdif-mp3";
case AF_FORMAT_S_TRUEHD: return "spdif-truehd";
}
return "??";
}
int af_fmt_seconds_to_bytes(int format, float seconds, int channels, int samplerate)
{
assert(!af_fmt_is_planar(format));
int bps = af_fmt_to_bytes(format);
int framelen = channels * bps;
int bytes = seconds * bps * samplerate;
if (bytes % framelen)
bytes += framelen - (bytes % framelen);
return bytes;
}
void af_fill_silence(void *dst, size_t bytes, int format)
{
memset(dst, af_fmt_is_unsigned(format) ? 0x80 : 0, bytes);
}
// Returns a "score" that serves as heuristic how lossy or hard a conversion is.
// If the formats are equal, 1024 is returned. If they are gravely incompatible
// (like s16<->ac3), INT_MIN is returned. If there is implied loss of precision
// (like s16->s8), a value <0 is returned.
static int af_format_conversion_score(int dst_format, int src_format)
{
if (dst_format == AF_FORMAT_UNKNOWN || src_format == AF_FORMAT_UNKNOWN)
return INT_MIN;
if (dst_format == src_format)
return 1024;
// Can't be normally converted
if (!af_fmt_is_pcm(dst_format) || !af_fmt_is_pcm(src_format))
return INT_MIN;
int score = 1024;
if (af_fmt_is_planar(dst_format) != af_fmt_is_planar(src_format))
score -= 1; // has to (de-)planarize
if (af_fmt_is_float(dst_format) != af_fmt_is_float(src_format)) {
int dst_bytes = af_fmt_to_bytes(dst_format);
if (af_fmt_is_float(dst_format)) {
// For int->float, consider a lower bound on the precision difference.
int bytes = (dst_bytes == 4 ? 3 : 6) - af_fmt_to_bytes(src_format);
if (bytes >= 0) {
score -= 8 * bytes; // excess precision
} else {
score += 1024 * (bytes - 1); // precision is lost (i.e. s32 -> float)
}
} else {
// float->int is the worst case. Penalize heavily and
// prefer highest bit depth int.
score -= 1048576 * (8 - dst_bytes);
}
score -= 512; // penalty for any float <-> int conversion
} else {
int bytes = af_fmt_to_bytes(dst_format) - af_fmt_to_bytes(src_format);
if (bytes > 0) {
score -= 8 * bytes; // has to add padding
} else if (bytes < 0) {
score += 1024 * (bytes - 1); // has to reduce bit depth
}
}
return score;
}
struct entry {
int fmt;
int score;
};
static int cmp_entry(const void *a, const void *b)
{
#define CMP_INT(a, b) (a > b ? 1 : (a < b ? -1 : 0))
return -CMP_INT(((struct entry *)a)->score, ((struct entry *)b)->score);
}
// Return a list of sample format compatible to src_format, sorted by order
// of preference. out_formats[0] will be src_format (as long as it's valid),
// and the list is terminated with 0 (AF_FORMAT_UNKNOWN).
// Keep in mind that this also returns formats with flipped interleaving
// (e.g. for s16, it returns [s16, s16p, ...]).
void af_get_best_sample_formats(int src_format, int out_formats[AF_FORMAT_COUNT])
{
int num = 0;
struct entry e[AF_FORMAT_COUNT];
for (int fmt = 1; fmt < AF_FORMAT_COUNT; fmt++) {
int score = af_format_conversion_score(fmt, src_format);
if (score > INT_MIN)
e[num++] = (struct entry){fmt, score};
}
qsort(e, num, sizeof(e[0]), cmp_entry);
for (int n = 0; n < num; n++)
out_formats[n] = e[n].fmt;
out_formats[num] = 0;
}
// Return the best match to src_samplerate from the list provided in the array
// *available, which must be terminated by 0, or itself NULL. If *available is
// empty or NULL, return a negative value. Exact match to src_samplerate is
// most preferred, followed by the lowest integer multiple, followed by the
// maximum of *available.
int af_select_best_samplerate(int src_samplerate, const int *available)
{
if (!available)
return -1;
int min_mult_rate = INT_MAX;
int max_rate = INT_MIN;
for (int i = 0; available[i]; i++) {
if (available[i] == src_samplerate)
return available[i];
if (!(available[i] % src_samplerate))
min_mult_rate = MPMIN(min_mult_rate, available[i]);
max_rate = MPMAX(max_rate, available[i]);
}
if (min_mult_rate < INT_MAX)
return min_mult_rate;
if (max_rate > INT_MIN)
return max_rate;
return -1;
}
// Return the number of samples that make up one frame in this format.
// You get the byte size by multiplying them with sample size and channel count.
int af_format_sample_alignment(int format)
{
switch (format) {
case AF_FORMAT_S_AAC: return 16384 / 4;
case AF_FORMAT_S_AC3: return 6144 / 4;
case AF_FORMAT_S_DTSHD: return 32768 / 16;
case AF_FORMAT_S_DTS: return 2048 / 4;
case AF_FORMAT_S_EAC3: return 24576 / 4;
case AF_FORMAT_S_MP3: return 4608 / 4;
case AF_FORMAT_S_TRUEHD: return 61440 / 16;
default: return 1;
}
}