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- /*
- * Musepack SV7 decoder
- * Copyright (c) 2006 Konstantin Shishkov
- *
- * This file is part of Libav.
- *
- * Libav 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.
- *
- * Libav 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 Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
- /**
- * @file
- * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
- * divided into 32 subbands.
- */
-
- #include "libavutil/channel_layout.h"
- #include "libavutil/internal.h"
- #include "libavutil/lfg.h"
-
- #include "avcodec.h"
- #include "bitstream.h"
- #include "internal.h"
- #include "mpegaudiodsp.h"
-
- #include "mpc.h"
- #include "mpc7data.h"
-
- #define BANDS 32
- #define SAMPLES_PER_BAND 36
- #define MPC_FRAME_SIZE (BANDS * SAMPLES_PER_BAND)
-
- static VLC scfi_vlc, dscf_vlc, hdr_vlc, quant_vlc[MPC7_QUANT_VLC_TABLES][2];
-
- static const uint16_t quant_offsets[MPC7_QUANT_VLC_TABLES*2 + 1] =
- {
- 0, 512, 1024, 1536, 2052, 2564, 3076, 3588, 4100, 4612, 5124,
- 5636, 6164, 6676, 7224
- };
-
-
- static av_cold int mpc7_decode_init(AVCodecContext * avctx)
- {
- int i, j;
- MPCContext *c = avctx->priv_data;
- BitstreamContext bc;
- LOCAL_ALIGNED_16(uint8_t, buf, [16]);
- static int vlc_initialized = 0;
-
- static VLC_TYPE scfi_table[1 << MPC7_SCFI_BITS][2];
- static VLC_TYPE dscf_table[1 << MPC7_DSCF_BITS][2];
- static VLC_TYPE hdr_table[1 << MPC7_HDR_BITS][2];
- static VLC_TYPE quant_tables[7224][2];
-
- /* Musepack SV7 is always stereo */
- if (avctx->channels != 2) {
- avpriv_request_sample(avctx, "%d channels", avctx->channels);
- return AVERROR_PATCHWELCOME;
- }
-
- if(avctx->extradata_size < 16){
- av_log(avctx, AV_LOG_ERROR, "Too small extradata size (%i)!\n", avctx->extradata_size);
- return -1;
- }
- memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
- av_lfg_init(&c->rnd, 0xDEADBEEF);
- ff_bswapdsp_init(&c->bdsp);
- ff_mpadsp_init(&c->mpadsp);
- c->bdsp.bswap_buf((uint32_t *) buf, (const uint32_t *) avctx->extradata, 4);
- ff_mpc_init();
- bitstream_init(&bc, buf, 128);
-
- c->IS = bitstream_read_bit(&bc);
- c->MSS = bitstream_read_bit(&bc);
- c->maxbands = bitstream_read(&bc, 6);
- if(c->maxbands >= BANDS){
- av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->maxbands);
- return -1;
- }
- bitstream_skip(&bc, 88);
- c->gapless = bitstream_read_bit(&bc);
- c->lastframelen = bitstream_read(&bc, 11);
- av_log(avctx, AV_LOG_DEBUG, "IS: %d, MSS: %d, TG: %d, LFL: %d, bands: %d\n",
- c->IS, c->MSS, c->gapless, c->lastframelen, c->maxbands);
- c->frames_to_skip = 0;
-
- avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
- avctx->channel_layout = AV_CH_LAYOUT_STEREO;
-
- if(vlc_initialized) return 0;
- av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
- scfi_vlc.table = scfi_table;
- scfi_vlc.table_allocated = 1 << MPC7_SCFI_BITS;
- if(init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
- &mpc7_scfi[1], 2, 1,
- &mpc7_scfi[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
- av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n");
- return -1;
- }
- dscf_vlc.table = dscf_table;
- dscf_vlc.table_allocated = 1 << MPC7_DSCF_BITS;
- if(init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
- &mpc7_dscf[1], 2, 1,
- &mpc7_dscf[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
- av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n");
- return -1;
- }
- hdr_vlc.table = hdr_table;
- hdr_vlc.table_allocated = 1 << MPC7_HDR_BITS;
- if(init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
- &mpc7_hdr[1], 2, 1,
- &mpc7_hdr[0], 2, 1, INIT_VLC_USE_NEW_STATIC)){
- av_log(avctx, AV_LOG_ERROR, "Cannot init HDR VLC\n");
- return -1;
- }
- for(i = 0; i < MPC7_QUANT_VLC_TABLES; i++){
- for(j = 0; j < 2; j++){
- quant_vlc[i][j].table = &quant_tables[quant_offsets[i*2 + j]];
- quant_vlc[i][j].table_allocated = quant_offsets[i*2 + j + 1] - quant_offsets[i*2 + j];
- if(init_vlc(&quant_vlc[i][j], 9, mpc7_quant_vlc_sizes[i],
- &mpc7_quant_vlc[i][j][1], 4, 2,
- &mpc7_quant_vlc[i][j][0], 4, 2, INIT_VLC_USE_NEW_STATIC)){
- av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j);
- return -1;
- }
- }
- }
- vlc_initialized = 1;
-
- return 0;
- }
-
- /**
- * Fill samples for given subband
- */
- static inline void idx_to_quant(MPCContext *c, BitstreamContext *bc, int idx, int *dst)
- {
- int i, i1, t;
- switch(idx){
- case -1:
- for(i = 0; i < SAMPLES_PER_BAND; i++){
- *dst++ = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
- }
- break;
- case 1:
- i1 = bitstream_read_bit(bc);
- for(i = 0; i < SAMPLES_PER_BAND/3; i++){
- t = bitstream_read_vlc(bc, quant_vlc[0][i1].table, 9, 2);
- *dst++ = mpc7_idx30[t];
- *dst++ = mpc7_idx31[t];
- *dst++ = mpc7_idx32[t];
- }
- break;
- case 2:
- i1 = bitstream_read_bit(bc);
- for(i = 0; i < SAMPLES_PER_BAND/2; i++){
- t = bitstream_read_vlc(bc, quant_vlc[1][i1].table, 9, 2);
- *dst++ = mpc7_idx50[t];
- *dst++ = mpc7_idx51[t];
- }
- break;
- case 3: case 4: case 5: case 6: case 7:
- i1 = bitstream_read_bit(bc);
- for(i = 0; i < SAMPLES_PER_BAND; i++)
- *dst++ = bitstream_read_vlc(bc, quant_vlc[idx - 1][i1].table, 9, 2) - mpc7_quant_vlc_off[idx - 1];
- break;
- case 8: case 9: case 10: case 11: case 12:
- case 13: case 14: case 15: case 16: case 17:
- t = (1 << (idx - 2)) - 1;
- for(i = 0; i < SAMPLES_PER_BAND; i++)
- *dst++ = bitstream_read(bc, idx - 1) - t;
- break;
- default: // case 0 and -2..-17
- return;
- }
- }
-
- static int get_scale_idx(BitstreamContext *bc, int ref)
- {
- int t = bitstream_read_vlc(bc, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
- if (t == 8)
- return bitstream_read(bc, 6);
- return av_clip_uintp2(ref + t, 7);
- }
-
- static int mpc7_decode_frame(AVCodecContext * avctx, void *data,
- int *got_frame_ptr, AVPacket *avpkt)
- {
- AVFrame *frame = data;
- const uint8_t *buf = avpkt->data;
- int buf_size;
- MPCContext *c = avctx->priv_data;
- BitstreamContext bc;
- int i, ch;
- int mb = -1;
- Band *bands = c->bands;
- int off, ret, last_frame, skip;
- int bits_used, bits_avail;
-
- memset(bands, 0, sizeof(*bands) * (c->maxbands + 1));
-
- buf_size = avpkt->size & ~3;
- if (buf_size <= 0) {
- av_log(avctx, AV_LOG_ERROR, "packet size is too small (%i bytes)\n",
- avpkt->size);
- return AVERROR_INVALIDDATA;
- }
- if (buf_size != avpkt->size) {
- av_log(avctx, AV_LOG_WARNING, "packet size is not a multiple of 4. "
- "extra bytes at the end will be skipped.\n");
- }
-
- skip = buf[0];
- last_frame = buf[1];
- buf += 4;
- buf_size -= 4;
-
- /* get output buffer */
- frame->nb_samples = last_frame ? c->lastframelen : MPC_FRAME_SIZE;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
- return ret;
- }
-
- av_fast_padded_malloc(&c->bits, &c->buf_size, buf_size);
- if (!c->bits)
- return AVERROR(ENOMEM);
- c->bdsp.bswap_buf((uint32_t *) c->bits, (const uint32_t *) buf,
- buf_size >> 2);
- bitstream_init8(&bc, c->bits, buf_size);
- bitstream_skip(&bc, skip);
-
- /* read subband indexes */
- for(i = 0; i <= c->maxbands; i++){
- for(ch = 0; ch < 2; ch++){
- int t = 4;
- if (i)
- t = bitstream_read_vlc(&bc, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
- if (t == 4)
- bands[i].res[ch] = bitstream_read(&bc, 4);
- else bands[i].res[ch] = av_clip(bands[i-1].res[ch] + t, 0, 17);
- }
-
- if(bands[i].res[0] || bands[i].res[1]){
- mb = i;
- if (c->MSS)
- bands[i].msf = bitstream_read_bit(&bc);
- }
- }
- /* get scale indexes coding method */
- for(i = 0; i <= mb; i++)
- for(ch = 0; ch < 2; ch++)
- if (bands[i].res[ch])
- bands[i].scfi[ch] = bitstream_read_vlc(&bc, scfi_vlc.table, MPC7_SCFI_BITS, 1);
- /* get scale indexes */
- for(i = 0; i <= mb; i++){
- for(ch = 0; ch < 2; ch++){
- if(bands[i].res[ch]){
- bands[i].scf_idx[ch][2] = c->oldDSCF[ch][i];
- bands[i].scf_idx[ch][0] = get_scale_idx(&bc, bands[i].scf_idx[ch][2]);
- switch(bands[i].scfi[ch]){
- case 0:
- bands[i].scf_idx[ch][1] = get_scale_idx(&bc, bands[i].scf_idx[ch][0]);
- bands[i].scf_idx[ch][2] = get_scale_idx(&bc, bands[i].scf_idx[ch][1]);
- break;
- case 1:
- bands[i].scf_idx[ch][1] = get_scale_idx(&bc, bands[i].scf_idx[ch][0]);
- bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1];
- break;
- case 2:
- bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
- bands[i].scf_idx[ch][2] = get_scale_idx(&bc, bands[i].scf_idx[ch][1]);
- break;
- case 3:
- bands[i].scf_idx[ch][2] = bands[i].scf_idx[ch][1] = bands[i].scf_idx[ch][0];
- break;
- }
- c->oldDSCF[ch][i] = bands[i].scf_idx[ch][2];
- }
- }
- }
- /* get quantizers */
- memset(c->Q, 0, sizeof(c->Q));
- off = 0;
- for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
- for(ch = 0; ch < 2; ch++)
- idx_to_quant(c, &bc, bands[i].res[ch], c->Q[ch] + off);
-
- ff_mpc_dequantize_and_synth(c, mb, (int16_t **)frame->extended_data, 2);
-
- bits_used = bitstream_tell(&bc);
- bits_avail = buf_size * 8;
- if (!last_frame && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))) {
- av_log(avctx, AV_LOG_ERROR, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
- return -1;
- }
- if(c->frames_to_skip){
- c->frames_to_skip--;
- *got_frame_ptr = 0;
- return avpkt->size;
- }
-
- *got_frame_ptr = 1;
-
- return avpkt->size;
- }
-
- static void mpc7_decode_flush(AVCodecContext *avctx)
- {
- MPCContext *c = avctx->priv_data;
-
- memset(c->oldDSCF, 0, sizeof(c->oldDSCF));
- c->frames_to_skip = 32;
- }
-
- static av_cold int mpc7_decode_close(AVCodecContext *avctx)
- {
- MPCContext *c = avctx->priv_data;
- av_freep(&c->bits);
- c->buf_size = 0;
- return 0;
- }
-
- AVCodec ff_mpc7_decoder = {
- .name = "mpc7",
- .long_name = NULL_IF_CONFIG_SMALL("Musepack SV7"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_MUSEPACK7,
- .priv_data_size = sizeof(MPCContext),
- .init = mpc7_decode_init,
- .close = mpc7_decode_close,
- .decode = mpc7_decode_frame,
- .flush = mpc7_decode_flush,
- .capabilities = AV_CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
- AV_SAMPLE_FMT_NONE },
- };
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