Files
ppsspp/ext/ffmpeg/libavcodec/mpeg4audio.c
T
Henrik RydgårdandClaude Opus 5.5 d0e3141f43 FFmpeg: Vendor what PPSSPP uses into ext/ffmpeg, drop the submodule
The ffmpeg submodule held prebuilt libraries for every platform next to the
sources (about 1.5 GB checked out), and nothing uses those any more. Only the
needed sources are copied in, from the submodule's commit, along with every
header and template they include. The path is new, so checking out commits on
either side of this works.

It's also cut down to what PPSSPP uses: the H.264, AAC, MP3, ATRAC3 and ATRAC3+
decoders, the H.264 parser, the demuxers as before, and the FFV1 encoder, AVI
muxer and file protocol for AVI dumping. config.h takes its CONFIG_ values from
FFmpeg 3.0.2's own configure run with just those enabled. The raw H.264 demuxer
stays even though nothing opens raw H.264: probing PSMF data uses it to identify
the video stream's codec, which otherwise comes out as none.

On top of that:
- The files nothing links (configure would build them, but they're never pulled
  out of the static library, going by the GCC link map) are left out: mostly
  libavutil's crypto and hashing and the alternative DCT/IDCT/FFT versions.
- libswresample isn't built, now that nothing uses it.
- The H.264 decoder only does 8-bit video, which is all the PSP has. The SPS
  parser rejects any other bit depth, and the DSP files no longer build their 9-
  to 16-bit variants, which the linker kept because the init functions pick one
  by bit depth at run time.

That's 160 files to compile instead of 277, and 391 of FFmpeg's vendored, about
5.8 MB. The README in ext/ffmpeg has what was done, the PPSSPP patches in the
copy, how to add a component, and the tradeoffs.

Co-Authored-By: Claude Opus 5.5 (1M context) <noreply@anthropic.com>
2026-10-25 22:27:54 +01:00

196 lines
6.4 KiB
C

/*
* MPEG-4 Audio common code
* Copyright (c) 2008 Baptiste Coudurier <baptiste.coudurier@free.fr>
* Copyright (c) 2009 Alex Converse <alex.converse@gmail.com>
*
* 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 "get_bits.h"
#include "put_bits.h"
#include "mpeg4audio.h"
/**
* Parse MPEG-4 audio configuration for ALS object type.
* @param[in] gb bit reader context
* @param[in] c MPEG4AudioConfig structure to fill
* @return on success 0 is returned, otherwise a value < 0
*/
static int parse_config_ALS(GetBitContext *gb, MPEG4AudioConfig *c)
{
if (get_bits_left(gb) < 112)
return -1;
if (get_bits_long(gb, 32) != MKBETAG('A','L','S','\0'))
return -1;
// override AudioSpecificConfig channel configuration and sample rate
// which are buggy in old ALS conformance files
c->sample_rate = get_bits_long(gb, 32);
// skip number of samples
skip_bits_long(gb, 32);
// read number of channels
c->chan_config = 0;
c->channels = get_bits(gb, 16) + 1;
return 0;
}
/* XXX: make sure to update the copies in the different encoders if you change
* this table */
const int avpriv_mpeg4audio_sample_rates[16] = {
96000, 88200, 64000, 48000, 44100, 32000,
24000, 22050, 16000, 12000, 11025, 8000, 7350
};
const uint8_t ff_mpeg4audio_channels[8] = {
0, 1, 2, 3, 4, 5, 6, 8
};
static inline int get_object_type(GetBitContext *gb)
{
int object_type = get_bits(gb, 5);
if (object_type == AOT_ESCAPE)
object_type = 32 + get_bits(gb, 6);
return object_type;
}
static inline int get_sample_rate(GetBitContext *gb, int *index)
{
*index = get_bits(gb, 4);
return *index == 0x0f ? get_bits(gb, 24) :
avpriv_mpeg4audio_sample_rates[*index];
}
int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf,
int bit_size, int sync_extension)
{
GetBitContext gb;
int specific_config_bitindex, ret;
if (bit_size <= 0)
return AVERROR_INVALIDDATA;
ret = init_get_bits(&gb, buf, bit_size);
if (ret < 0)
return ret;
c->object_type = get_object_type(&gb);
c->sample_rate = get_sample_rate(&gb, &c->sampling_index);
c->chan_config = get_bits(&gb, 4);
if (c->chan_config < FF_ARRAY_ELEMS(ff_mpeg4audio_channels))
c->channels = ff_mpeg4audio_channels[c->chan_config];
c->sbr = -1;
c->ps = -1;
if (c->object_type == AOT_SBR || (c->object_type == AOT_PS &&
// check for W6132 Annex YYYY draft MP3onMP4
!(show_bits(&gb, 3) & 0x03 && !(show_bits(&gb, 9) & 0x3F)))) {
if (c->object_type == AOT_PS)
c->ps = 1;
c->ext_object_type = AOT_SBR;
c->sbr = 1;
c->ext_sample_rate = get_sample_rate(&gb, &c->ext_sampling_index);
c->object_type = get_object_type(&gb);
if (c->object_type == AOT_ER_BSAC)
c->ext_chan_config = get_bits(&gb, 4);
} else {
c->ext_object_type = AOT_NULL;
c->ext_sample_rate = 0;
}
specific_config_bitindex = get_bits_count(&gb);
if (c->object_type == AOT_ALS) {
skip_bits(&gb, 5);
if (show_bits_long(&gb, 24) != MKBETAG('\0','A','L','S'))
skip_bits_long(&gb, 24);
specific_config_bitindex = get_bits_count(&gb);
if (parse_config_ALS(&gb, c))
return -1;
}
if (c->ext_object_type != AOT_SBR && sync_extension) {
while (get_bits_left(&gb) > 15) {
if (show_bits(&gb, 11) == 0x2b7) { // sync extension
get_bits(&gb, 11);
c->ext_object_type = get_object_type(&gb);
if (c->ext_object_type == AOT_SBR && (c->sbr = get_bits1(&gb)) == 1) {
c->ext_sample_rate = get_sample_rate(&gb, &c->ext_sampling_index);
if (c->ext_sample_rate == c->sample_rate)
c->sbr = -1;
}
if (get_bits_left(&gb) > 11 && get_bits(&gb, 11) == 0x548)
c->ps = get_bits1(&gb);
break;
} else
get_bits1(&gb); // skip 1 bit
}
}
//PS requires SBR
if (!c->sbr)
c->ps = 0;
//Limit implicit PS to the HE-AACv2 Profile
if ((c->ps == -1 && c->object_type != AOT_AAC_LC) || c->channels & ~0x01)
c->ps = 0;
return specific_config_bitindex;
}
static av_always_inline unsigned int copy_bits(PutBitContext *pb,
GetBitContext *gb,
int bits)
{
unsigned int el = get_bits(gb, bits);
put_bits(pb, bits, el);
return el;
}
int avpriv_copy_pce_data(PutBitContext *pb, GetBitContext *gb)
{
int five_bit_ch, four_bit_ch, comment_size, bits;
int offset = put_bits_count(pb);
copy_bits(pb, gb, 10); //Tag, Object Type, Frequency
five_bit_ch = copy_bits(pb, gb, 4); //Front
five_bit_ch += copy_bits(pb, gb, 4); //Side
five_bit_ch += copy_bits(pb, gb, 4); //Back
four_bit_ch = copy_bits(pb, gb, 2); //LFE
four_bit_ch += copy_bits(pb, gb, 3); //Data
five_bit_ch += copy_bits(pb, gb, 4); //Coupling
if (copy_bits(pb, gb, 1)) //Mono Mixdown
copy_bits(pb, gb, 4);
if (copy_bits(pb, gb, 1)) //Stereo Mixdown
copy_bits(pb, gb, 4);
if (copy_bits(pb, gb, 1)) //Matrix Mixdown
copy_bits(pb, gb, 3);
for (bits = five_bit_ch*5+four_bit_ch*4; bits > 16; bits -= 16)
copy_bits(pb, gb, 16);
if (bits)
copy_bits(pb, gb, bits);
avpriv_align_put_bits(pb);
align_get_bits(gb);
comment_size = copy_bits(pb, gb, 8);
for (; comment_size > 0; comment_size--)
copy_bits(pb, gb, 8);
return put_bits_count(pb) - offset;
}