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ppsspp/ext/ffmpeg/libavcodec/atrac3plusdec.c
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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

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/*
* ATRAC3+ compatible decoder
*
* Copyright (c) 2010-2013 Maxim Poliakovski
*
* 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
*/
/**
* @file
* Sony ATRAC3+ compatible decoder.
*
* Container formats used to store its data:
* RIFF WAV (.at3) and Sony OpenMG (.oma, .aa3).
*
* Technical description of this codec can be found here:
* http://wiki.multimedia.cx/index.php?title=ATRAC3plus
*
* Kudos to Benjamin Larsson and Michael Karcher
* for their precious technical help!
*/
#include <stdint.h>
#include <string.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "get_bits.h"
#include "internal.h"
#include "atrac.h"
#include "atrac3plus.h"
typedef struct ATRAC3PContext {
GetBitContext gb;
AVFloatDSPContext *fdsp;
DECLARE_ALIGNED(32, float, samples)[2][ATRAC3P_FRAME_SAMPLES]; ///< quantized MDCT spectrum
DECLARE_ALIGNED(32, float, mdct_buf)[2][ATRAC3P_FRAME_SAMPLES]; ///< output of the IMDCT
DECLARE_ALIGNED(32, float, time_buf)[2][ATRAC3P_FRAME_SAMPLES]; ///< output of the gain compensation
DECLARE_ALIGNED(32, float, outp_buf)[2][ATRAC3P_FRAME_SAMPLES];
AtracGCContext gainc_ctx; ///< gain compensation context
FFTContext mdct_ctx;
FFTContext ipqf_dct_ctx; ///< IDCT context used by IPQF
Atrac3pChanUnitCtx *ch_units; ///< global channel units
int num_channel_blocks; ///< number of channel blocks
uint8_t channel_blocks[5]; ///< channel configuration descriptor
uint64_t my_channel_layout; ///< current channel layout
} ATRAC3PContext;
static av_cold int atrac3p_decode_close(AVCodecContext *avctx)
{
ATRAC3PContext *ctx = avctx->priv_data;
av_freep(&ctx->ch_units);
av_freep(&ctx->fdsp);
ff_mdct_end(&ctx->mdct_ctx);
ff_mdct_end(&ctx->ipqf_dct_ctx);
return 0;
}
static av_cold int set_channel_params(ATRAC3PContext *ctx,
AVCodecContext *avctx)
{
memset(ctx->channel_blocks, 0, sizeof(ctx->channel_blocks));
switch (avctx->channels) {
case 1:
if (avctx->channel_layout != AV_CH_FRONT_LEFT)
avctx->channel_layout = AV_CH_LAYOUT_MONO;
ctx->num_channel_blocks = 1;
ctx->channel_blocks[0] = CH_UNIT_MONO;
break;
case 2:
avctx->channel_layout = AV_CH_LAYOUT_STEREO;
ctx->num_channel_blocks = 1;
ctx->channel_blocks[0] = CH_UNIT_STEREO;
break;
case 3:
avctx->channel_layout = AV_CH_LAYOUT_SURROUND;
ctx->num_channel_blocks = 2;
ctx->channel_blocks[0] = CH_UNIT_STEREO;
ctx->channel_blocks[1] = CH_UNIT_MONO;
break;
case 4:
avctx->channel_layout = AV_CH_LAYOUT_4POINT0;
ctx->num_channel_blocks = 3;
ctx->channel_blocks[0] = CH_UNIT_STEREO;
ctx->channel_blocks[1] = CH_UNIT_MONO;
ctx->channel_blocks[2] = CH_UNIT_MONO;
break;
case 6:
avctx->channel_layout = AV_CH_LAYOUT_5POINT1_BACK;
ctx->num_channel_blocks = 4;
ctx->channel_blocks[0] = CH_UNIT_STEREO;
ctx->channel_blocks[1] = CH_UNIT_MONO;
ctx->channel_blocks[2] = CH_UNIT_STEREO;
ctx->channel_blocks[3] = CH_UNIT_MONO;
break;
case 7:
avctx->channel_layout = AV_CH_LAYOUT_6POINT1_BACK;
ctx->num_channel_blocks = 5;
ctx->channel_blocks[0] = CH_UNIT_STEREO;
ctx->channel_blocks[1] = CH_UNIT_MONO;
ctx->channel_blocks[2] = CH_UNIT_STEREO;
ctx->channel_blocks[3] = CH_UNIT_MONO;
ctx->channel_blocks[4] = CH_UNIT_MONO;
break;
case 8:
avctx->channel_layout = AV_CH_LAYOUT_7POINT1;
ctx->num_channel_blocks = 5;
ctx->channel_blocks[0] = CH_UNIT_STEREO;
ctx->channel_blocks[1] = CH_UNIT_MONO;
ctx->channel_blocks[2] = CH_UNIT_STEREO;
ctx->channel_blocks[3] = CH_UNIT_STEREO;
ctx->channel_blocks[4] = CH_UNIT_MONO;
break;
default:
av_log(avctx, AV_LOG_ERROR,
"Unsupported channel count: %d!\n", avctx->channels);
return AVERROR_INVALIDDATA;
}
return 0;
}
static av_cold int atrac3p_decode_init(AVCodecContext *avctx)
{
ATRAC3PContext *ctx = avctx->priv_data;
int i, ch, ret;
if (!avctx->block_align) {
av_log(avctx, AV_LOG_ERROR, "block_align is not set\n");
return AVERROR(EINVAL);
}
ff_atrac3p_init_vlcs();
/* initialize IPQF */
ff_mdct_init(&ctx->ipqf_dct_ctx, 5, 1, 32.0 / 32768.0);
ff_atrac3p_init_imdct(avctx, &ctx->mdct_ctx);
ff_atrac_init_gain_compensation(&ctx->gainc_ctx, 6, 2);
ff_atrac3p_init_wave_synth();
if ((ret = set_channel_params(ctx, avctx)) < 0)
return ret;
ctx->my_channel_layout = avctx->channel_layout;
ctx->ch_units = av_mallocz_array(ctx->num_channel_blocks, sizeof(*ctx->ch_units));
ctx->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
if (!ctx->ch_units || !ctx->fdsp) {
atrac3p_decode_close(avctx);
return AVERROR(ENOMEM);
}
for (i = 0; i < ctx->num_channel_blocks; i++) {
for (ch = 0; ch < 2; ch++) {
ctx->ch_units[i].channels[ch].ch_num = ch;
ctx->ch_units[i].channels[ch].wnd_shape = &ctx->ch_units[i].channels[ch].wnd_shape_hist[0][0];
ctx->ch_units[i].channels[ch].wnd_shape_prev = &ctx->ch_units[i].channels[ch].wnd_shape_hist[1][0];
ctx->ch_units[i].channels[ch].gain_data = &ctx->ch_units[i].channels[ch].gain_data_hist[0][0];
ctx->ch_units[i].channels[ch].gain_data_prev = &ctx->ch_units[i].channels[ch].gain_data_hist[1][0];
ctx->ch_units[i].channels[ch].tones_info = &ctx->ch_units[i].channels[ch].tones_info_hist[0][0];
ctx->ch_units[i].channels[ch].tones_info_prev = &ctx->ch_units[i].channels[ch].tones_info_hist[1][0];
}
ctx->ch_units[i].waves_info = &ctx->ch_units[i].wave_synth_hist[0];
ctx->ch_units[i].waves_info_prev = &ctx->ch_units[i].wave_synth_hist[1];
}
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
return 0;
}
static void decode_residual_spectrum(Atrac3pChanUnitCtx *ctx,
float out[2][ATRAC3P_FRAME_SAMPLES],
int num_channels,
AVCodecContext *avctx)
{
int i, sb, ch, qu, nspeclines, RNG_index;
float *dst, q;
int16_t *src;
/* calculate RNG table index for each subband */
int sb_RNG_index[ATRAC3P_SUBBANDS] = { 0 };
if (ctx->mute_flag) {
for (ch = 0; ch < num_channels; ch++)
memset(out[ch], 0, ATRAC3P_FRAME_SAMPLES * sizeof(*out[ch]));
return;
}
for (qu = 0, RNG_index = 0; qu < ctx->used_quant_units; qu++)
RNG_index += ctx->channels[0].qu_sf_idx[qu] +
ctx->channels[1].qu_sf_idx[qu];
for (sb = 0; sb < ctx->num_coded_subbands; sb++, RNG_index += 128)
sb_RNG_index[sb] = RNG_index & 0x3FC;
/* inverse quant and power compensation */
for (ch = 0; ch < num_channels; ch++) {
/* clear channel's residual spectrum */
memset(out[ch], 0, ATRAC3P_FRAME_SAMPLES * sizeof(*out[ch]));
for (qu = 0; qu < ctx->used_quant_units; qu++) {
src = &ctx->channels[ch].spectrum[ff_atrac3p_qu_to_spec_pos[qu]];
dst = &out[ch][ff_atrac3p_qu_to_spec_pos[qu]];
nspeclines = ff_atrac3p_qu_to_spec_pos[qu + 1] -
ff_atrac3p_qu_to_spec_pos[qu];
if (ctx->channels[ch].qu_wordlen[qu] > 0) {
q = ff_atrac3p_sf_tab[ctx->channels[ch].qu_sf_idx[qu]] *
ff_atrac3p_mant_tab[ctx->channels[ch].qu_wordlen[qu]];
for (i = 0; i < nspeclines; i++)
dst[i] = src[i] * q;
}
}
for (sb = 0; sb < ctx->num_coded_subbands; sb++)
ff_atrac3p_power_compensation(ctx, ch, &out[ch][0],
sb_RNG_index[sb], sb);
}
if (ctx->unit_type == CH_UNIT_STEREO) {
for (sb = 0; sb < ctx->num_coded_subbands; sb++) {
if (ctx->swap_channels[sb]) {
for (i = 0; i < ATRAC3P_SUBBAND_SAMPLES; i++)
FFSWAP(float, out[0][sb * ATRAC3P_SUBBAND_SAMPLES + i],
out[1][sb * ATRAC3P_SUBBAND_SAMPLES + i]);
}
/* flip coefficients' sign if requested */
if (ctx->negate_coeffs[sb])
for (i = 0; i < ATRAC3P_SUBBAND_SAMPLES; i++)
out[1][sb * ATRAC3P_SUBBAND_SAMPLES + i] = -(out[1][sb * ATRAC3P_SUBBAND_SAMPLES + i]);
}
}
}
static void reconstruct_frame(ATRAC3PContext *ctx, Atrac3pChanUnitCtx *ch_unit,
int num_channels, AVCodecContext *avctx)
{
int ch, sb;
for (ch = 0; ch < num_channels; ch++) {
for (sb = 0; sb < ch_unit->num_subbands; sb++) {
/* inverse transform and windowing */
ff_atrac3p_imdct(ctx->fdsp, &ctx->mdct_ctx,
&ctx->samples[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
&ctx->mdct_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
(ch_unit->channels[ch].wnd_shape_prev[sb] << 1) +
ch_unit->channels[ch].wnd_shape[sb], sb);
/* gain compensation and overlapping */
ff_atrac_gain_compensation(&ctx->gainc_ctx,
&ctx->mdct_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
&ch_unit->prev_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES],
&ch_unit->channels[ch].gain_data_prev[sb],
&ch_unit->channels[ch].gain_data[sb],
ATRAC3P_SUBBAND_SAMPLES,
&ctx->time_buf[ch][sb * ATRAC3P_SUBBAND_SAMPLES]);
}
/* zero unused subbands in both output and overlapping buffers */
memset(&ch_unit->prev_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES],
0,
(ATRAC3P_SUBBANDS - ch_unit->num_subbands) *
ATRAC3P_SUBBAND_SAMPLES *
sizeof(ch_unit->prev_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES]));
memset(&ctx->time_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES],
0,
(ATRAC3P_SUBBANDS - ch_unit->num_subbands) *
ATRAC3P_SUBBAND_SAMPLES *
sizeof(ctx->time_buf[ch][ch_unit->num_subbands * ATRAC3P_SUBBAND_SAMPLES]));
/* resynthesize and add tonal signal */
if (ch_unit->waves_info->tones_present ||
ch_unit->waves_info_prev->tones_present) {
for (sb = 0; sb < ch_unit->num_subbands; sb++)
if (ch_unit->channels[ch].tones_info[sb].num_wavs ||
ch_unit->channels[ch].tones_info_prev[sb].num_wavs) {
ff_atrac3p_generate_tones(ch_unit, ctx->fdsp, ch, sb,
&ctx->time_buf[ch][sb * 128]);
}
}
/* subband synthesis and acoustic signal output */
ff_atrac3p_ipqf(&ctx->ipqf_dct_ctx, &ch_unit->ipqf_ctx[ch],
&ctx->time_buf[ch][0], &ctx->outp_buf[ch][0]);
}
/* swap window shape and gain control buffers. */
for (ch = 0; ch < num_channels; ch++) {
FFSWAP(uint8_t *, ch_unit->channels[ch].wnd_shape,
ch_unit->channels[ch].wnd_shape_prev);
FFSWAP(AtracGainInfo *, ch_unit->channels[ch].gain_data,
ch_unit->channels[ch].gain_data_prev);
FFSWAP(Atrac3pWavesData *, ch_unit->channels[ch].tones_info,
ch_unit->channels[ch].tones_info_prev);
}
FFSWAP(Atrac3pWaveSynthParams *, ch_unit->waves_info, ch_unit->waves_info_prev);
}
static int atrac3p_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
ATRAC3PContext *ctx = avctx->priv_data;
AVFrame *frame = data;
int i, ret, ch_unit_id, ch_block = 0, out_ch_index = 0, channels_to_process;
float **samples_p = (float **)frame->extended_data;
frame->nb_samples = ATRAC3P_FRAME_SAMPLES;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
if ((ret = init_get_bits8(&ctx->gb, avpkt->data, avpkt->size)) < 0)
return ret;
if (get_bits1(&ctx->gb)) {
av_log(avctx, AV_LOG_ERROR, "Invalid start bit!\n");
return AVERROR_INVALIDDATA;
}
while (get_bits_left(&ctx->gb) >= 2 &&
(ch_unit_id = get_bits(&ctx->gb, 2)) != CH_UNIT_TERMINATOR) {
if (ch_unit_id == CH_UNIT_EXTENSION) {
avpriv_report_missing_feature(avctx, "Channel unit extension");
return AVERROR_PATCHWELCOME;
}
if (ch_block >= ctx->num_channel_blocks ||
ctx->channel_blocks[ch_block] != ch_unit_id) {
av_log(avctx, AV_LOG_ERROR,
"Frame data doesn't match channel configuration!\n");
return AVERROR_INVALIDDATA;
}
ctx->ch_units[ch_block].unit_type = ch_unit_id;
channels_to_process = ch_unit_id + 1;
if ((ret = ff_atrac3p_decode_channel_unit(&ctx->gb,
&ctx->ch_units[ch_block],
channels_to_process,
avctx)) < 0)
return ret;
decode_residual_spectrum(&ctx->ch_units[ch_block], ctx->samples,
channels_to_process, avctx);
reconstruct_frame(ctx, &ctx->ch_units[ch_block],
channels_to_process, avctx);
for (i = 0; i < channels_to_process; i++)
memcpy(samples_p[out_ch_index + i], ctx->outp_buf[i],
ATRAC3P_FRAME_SAMPLES * sizeof(**samples_p));
ch_block++;
out_ch_index += channels_to_process;
}
*got_frame_ptr = 1;
return FFMIN(avctx->block_align, avpkt->size);
}
AVCodec ff_atrac3p_decoder = {
.name = "atrac3plus",
.long_name = NULL_IF_CONFIG_SMALL("ATRAC3+ (Adaptive TRansform Acoustic Coding 3+)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_ATRAC3P,
.capabilities = AV_CODEC_CAP_DR1,
.priv_data_size = sizeof(ATRAC3PContext),
.init = atrac3p_decode_init,
.close = atrac3p_decode_close,
.decode = atrac3p_decode_frame,
};
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