e4de71677f
* qatar/master: aac_latm: reconfigure decoder on audio specific config changes latmdec: fix audio specific config parsing Add avcodec_decode_audio4(). avcodec: change number of plane pointers from 4 to 8 at next major bump. Update developers documentation with coding conventions. svq1dec: avoid undefined get_bits(0) call ARM: h264dsp_neon cosmetics ARM: make some NEON macros reusable Do not memcpy raw video frames when using null muxer fate: update asf seektest vp8: flush buffers on size changes. doc: improve general documentation for MacOSX asf: use packet dts as approximation of pts asf: do not call av_read_frame rtsp: Initialize the media_type_mask in the rtp guessing demuxer Cleaned up alacenc.c Conflicts: doc/APIchanges doc/developer.texi libavcodec/8svx.c libavcodec/aacdec.c libavcodec/ac3dec.c libavcodec/avcodec.h libavcodec/nellymoserdec.c libavcodec/tta.c libavcodec/utils.c libavcodec/version.h libavcodec/wmadec.c libavformat/asfdec.c tests/ref/seek/lavf_asf Merged-by: Michael Niedermayer <michaelni@gmx.at>
324 lines
11 KiB
C
324 lines
11 KiB
C
/*
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* Musepack SV7 decoder
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* Copyright (c) 2006 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
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* divided into 32 subbands.
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*/
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#include "libavutil/lfg.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "mpegaudiodsp.h"
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#include "libavutil/audioconvert.h"
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#include "mpc.h"
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#include "mpc7data.h"
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#define BANDS 32
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#define SAMPLES_PER_BAND 36
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#define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
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static VLC scfi_vlc, dscf_vlc, hdr_vlc, quant_vlc[MPC7_QUANT_VLC_TABLES][2];
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static const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES*2 + 1] =
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{
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0, 512, 1024, 1536, 2052, 2564, 3076, 3588, 4100, 4612, 5124,
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5636, 6164, 6676, 7224
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};
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static av_cold int mpc7_decode_init(AVCodecContext * avctx)
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{
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int i, j;
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MPCContext *c = avctx->priv_data;
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GetBitContext gb;
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uint8_t buf[16];
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static int vlc_initialized = 0;
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static VLC_TYPE scfi_table[1 << MPC7_SCFI_BITS][2];
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static VLC_TYPE dscf_table[1 << MPC7_DSCF_BITS][2];
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static VLC_TYPE hdr_table[1 << MPC7_HDR_BITS][2];
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static VLC_TYPE quant_tables[7224][2];
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/* Musepack SV7 is always stereo */
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if (avctx->channels != 2) {
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av_log_ask_for_sample(avctx, "Unsupported number of channels: %d\n",
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avctx->channels);
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return AVERROR_PATCHWELCOME;
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}
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if(avctx->extradata_size < 16){
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av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
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return -1;
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}
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memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
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av_lfg_init(&c->rnd, 0xDEADBEEF);
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dsputil_init(&c->dsp, avctx);
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ff_mpadsp_init(&c->mpadsp);
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c->dsp.bswap_buf((uint32_t*)buf, (const uint32_t*)avctx->extradata, 4);
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ff_mpc_init();
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init_get_bits(&gb, buf, 128);
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c->IS = get_bits1(&gb);
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c->MSS = get_bits1(&gb);
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c->maxbands = get_bits(&gb, 6);
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if(c->maxbands >= BANDS){
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av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
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return -1;
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}
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skip_bits_long(&gb, 88);
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c->gapless = get_bits1(&gb);
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c->lastframelen = get_bits(&gb, 11);
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av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
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c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
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c->frames_to_skip = 0;
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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avctx->channel_layout = AV_CH_LAYOUT_STEREO;
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if(vlc_initialized) return 0;
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av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
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scfi_vlc.table = scfi_table;
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scfi_vlc.table_allocated = 1 << MPC7_SCFI_BITS;
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if(init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
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&mpc7_scfi[1], 2, 1,
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&mpc7_scfi[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n");
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return -1;
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}
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dscf_vlc.table = dscf_table;
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dscf_vlc.table_allocated = 1 << MPC7_DSCF_BITS;
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if(init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
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&mpc7_dscf[1], 2, 1,
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&mpc7_dscf[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n");
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return -1;
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}
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hdr_vlc.table = hdr_table;
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hdr_vlc.table_allocated = 1 << MPC7_HDR_BITS;
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if(init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
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&mpc7_hdr[1], 2, 1,
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&mpc7_hdr[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init HDR VLC\n");
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return -1;
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}
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for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){
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for(j = 0; j < 2; j++){
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quant_vlc[i][j].table = &quant_tables[quant_offsets[i*2 + j]];
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quant_vlc[i][j].table_allocated = quant_offsets[i*2 + j + 1] - quant_offsets[i*2 + j];
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if(init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
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&mpc7_quant_vlc[i][j][1], 4, 2,
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&mpc7_quant_vlc[i][j][0], 4, 2, INIT_VLC_USE_NEW_STATIC)){
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av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j);
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return -1;
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}
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}
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}
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vlc_initialized = 1;
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avcodec_get_frame_defaults(&c->frame);
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avctx->coded_frame = &c->frame;
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return 0;
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}
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/**
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* Fill samples for given subband
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*/
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static inline void idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
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{
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int i, i1, t;
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switch(idx){
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case -1:
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for(i = 0; i < SAMPLES_PER_BAND; i++){
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*dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
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}
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break;
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case 1:
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i1 = get_bits1(gb);
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for(i = 0; i < SAMPLES_PER_BAND/3; i++){
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t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
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*dst++ = mpc7_idx30[t];
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*dst++ = mpc7_idx31[t];
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*dst++ = mpc7_idx32[t];
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}
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break;
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case 2:
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i1 = get_bits1(gb);
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for(i = 0; i < SAMPLES_PER_BAND/2; i++){
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t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
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*dst++ = mpc7_idx50[t];
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*dst++ = mpc7_idx51[t];
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}
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break;
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case 3: case 4: case 5: case 6: case 7:
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i1 = get_bits1(gb);
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for(i = 0; i < SAMPLES_PER_BAND; i++)
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*dst++ = get_vlc2(gb, quant_vlc[idx-1][i1].table, 9, 2) - mpc7_quant_vlc_off[idx-1];
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break;
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case 8: case 9: case 10: case 11: case 12:
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case 13: case 14: case 15: case 16: case 17:
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t = (1 << (idx - 2)) - 1;
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for(i = 0; i < SAMPLES_PER_BAND; i++)
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*dst++ = get_bits(gb, idx - 1) - t;
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break;
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default: // case 0 and -2..-17
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return;
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}
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}
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static int get_scale_idx(GetBitContext *gb, int ref)
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{
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int t = get_vlc2(gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
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if (t == 8)
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return get_bits(gb, 6);
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return ref + t;
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}
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static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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MPCContext *c = avctx->priv_data;
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GetBitContext gb;
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uint8_t *bits;
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int i, ch;
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int mb = -1;
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Band *bands = c->bands;
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int off, ret;
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int bits_used, bits_avail;
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memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
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if(buf_size <= 4){
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av_log(avctx, AV_LOG_ERROR, "Too small buffer passed (%i bytes)\n", buf_size);
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return AVERROR(EINVAL);
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}
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/* get output buffer */
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c->frame.nb_samples = buf[1] ? c->lastframelen : MPC_FRAME_SIZE;
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if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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bits = av_malloc(((buf_size - 1) & ~3) + FF_INPUT_BUFFER_PADDING_SIZE);
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c->dsp.bswap_buf((uint32_t*)bits, (const uint32_t*)(buf + 4), (buf_size - 4) >> 2);
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init_get_bits(&gb, bits, (buf_size - 4)* 8);
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skip_bits_long(&gb, buf[0]);
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/* read subband indexes */
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for(i = 0; i <= c->maxbands; i++){
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for(ch = 0; ch < 2; ch++){
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int t = 4;
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if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
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if(t == 4) bands[i].res[ch] = get_bits(&gb, 4);
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else bands[i].res[ch] = bands[i-1].res[ch] + t;
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}
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if(bands[i].res[0] || bands[i].res[1]){
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mb = i;
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if(c->MSS) bands[i].msf = get_bits1(&gb);
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}
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}
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/* get scale indexes coding method */
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for(i = 0; i <= mb; i++)
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for(ch = 0; ch < 2; ch++)
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if(bands[i].res[ch]) bands[i].scfi[ch] = get_vlc2(&gb, scfi_vlc.table, MPC7_SCFI_BITS, 1);
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/* get scale indexes */
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for(i = 0; i <= mb; i++){
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for(ch = 0; ch < 2; ch++){
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if(bands[i].res[ch]){
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bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
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bands[i].scf_idx[ch][0] = get_scale_idx(&gb, bands[i].scf_idx[ch][2]);
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switch(bands[i].scfi[ch]){
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case 0:
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bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
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bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
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break;
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case 1:
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bands[i].scf_idx[ch][1] = get_scale_idx(&gb, bands[i].scf_idx[ch][0]);
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bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
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break;
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case 2:
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bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
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bands[i].scf_idx[ch][2] = get_scale_idx(&gb, bands[i].scf_idx[ch][1]);
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break;
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case 3:
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bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
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break;
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}
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c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
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}
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}
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}
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/* get quantizers */
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memset(c->Q, 0, sizeof(c->Q));
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off = 0;
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for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
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for(ch = 0; ch < 2; ch++)
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idx_to_quant(c, &gb, bands[i].res[ch], c->Q[ch] + off);
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ff_mpc_dequantize_and_synth(c, mb, c->frame.data[0], 2);
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av_free(bits);
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bits_used = get_bits_count(&gb);
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bits_avail = (buf_size - 4) * 8;
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if(!buf[1] && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))){
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av_log(NULL,0, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
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return -1;
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}
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if(c->frames_to_skip){
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c->frames_to_skip--;
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*got_frame_ptr = 0;
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return buf_size;
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}
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*got_frame_ptr = 1;
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*(AVFrame *)data = c->frame;
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return buf_size;
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}
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static void mpc7_decode_flush(AVCodecContext *avctx)
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{
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MPCContext *c = avctx->priv_data;
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memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
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c->frames_to_skip = 32;
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}
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AVCodec ff_mpc7_decoder = {
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.name = "mpc7",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_MUSEPACK7,
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.priv_data_size = sizeof(MPCContext),
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.init = mpc7_decode_init,
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.decode = mpc7_decode_frame,
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.flush = mpc7_decode_flush,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
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};
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