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authorKostya Shishkov <kostya.shishkov@gmail.com>2006-12-24 04:51:43 +0000
committerKostya Shishkov <kostya.shishkov@gmail.com>2006-12-24 04:51:43 +0000
commit185c7b6b2663ca22b5ae85e51b6ff5deacf5c038 (patch)
tree1f5550581c04ecfaff75909d37a362d60e925eac /libavcodec/mpc.c
parentf8aa696f9fde35ea7d5a42ef3fb65a5771fe02cc (diff)
downloadffmpeg-185c7b6b2663ca22b5ae85e51b6ff5deacf5c038.tar.gz
Musepack SV7 decoding support
Originally committed as revision 7375 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/mpc.c')
-rw-r--r--libavcodec/mpc.c346
1 files changed, 346 insertions, 0 deletions
diff --git a/libavcodec/mpc.c b/libavcodec/mpc.c
new file mode 100644
index 0000000000..dcecb0ee5a
--- /dev/null
+++ b/libavcodec/mpc.c
@@ -0,0 +1,346 @@
+/*
+ * Musepack decoder
+ * Copyright (c) 2006 Konstantin Shishkov
+ *
+ * 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 mpc.c Musepack decoder
+ * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
+ * divided into 32 subbands.
+ */
+
+#include "avcodec.h"
+#include "bitstream.h"
+#include "dsputil.h"
+
+#ifdef CONFIG_MPEGAUDIO_HP
+#define USE_HIGHPRECISION
+#endif
+#include "mpegaudio.h"
+
+#include "mpcdata.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 DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
+
+typedef struct {
+ DSPContext dsp;
+ int IS, MSS, gapless;
+ int lastframelen, bands;
+ int oldDSCF[2][BANDS];
+ int rnd;
+ /* for synthesis */
+ DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
+ int synth_buf_offset[MPA_MAX_CHANNELS];
+ DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT]);
+} MPCContext;
+
+/** Subband structure - hold all variables for each subband */
+typedef struct {
+ int msf; ///< mid-stereo flag
+ int res[2];
+ int scfi[2];
+ int scf_idx[2][3];
+ int Q[2];
+}Band;
+
+static int mpc7_decode_init(AVCodecContext * avctx)
+{
+ int i, j;
+ MPCContext *c = avctx->priv_data;
+ GetBitContext gb;
+ uint8_t buf[16];
+ float f1=1.20050805774840750476 * 256;
+ static int vlc_inited = 0;
+
+ 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));
+ c->rnd = 0xDEADBEEF;
+ dsputil_init(&c->dsp, avctx);
+ c->dsp.bswap_buf(buf, avctx->extradata, 4);
+ ff_mpa_synth_init(mpa_window);
+ init_get_bits(&gb, buf, 128);
+
+ c->IS = get_bits1(&gb);
+ c->MSS = get_bits1(&gb);
+ c->bands = get_bits(&gb, 6);
+ if(c->bands >= BANDS){
+ av_log(avctx, AV_LOG_ERROR, "Too many bands: %i\n", c->bands);
+ return -1;
+ }
+ skip_bits(&gb, 88);
+ c->gapless = get_bits1(&gb);
+ c->lastframelen = get_bits(&gb, 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->bands);
+
+ if(vlc_inited) return 0;
+ av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
+ if(init_vlc(&scfi_vlc, MPC7_SCFI_BITS, MPC7_SCFI_SIZE,
+ &mpc7_scfi[1], 2, 1,
+ &mpc7_scfi[0], 2, 1, INIT_VLC_USE_STATIC)){
+ av_log(avctx, AV_LOG_ERROR, "Cannot init SCFI VLC\n");
+ return -1;
+ }
+ if(init_vlc(&dscf_vlc, MPC7_DSCF_BITS, MPC7_DSCF_SIZE,
+ &mpc7_dscf[1], 2, 1,
+ &mpc7_dscf[0], 2, 1, INIT_VLC_USE_STATIC)){
+ av_log(avctx, AV_LOG_ERROR, "Cannot init DSCF VLC\n");
+ return -1;
+ }
+ if(init_vlc(&hdr_vlc, MPC7_HDR_BITS, MPC7_HDR_SIZE,
+ &mpc7_hdr[1], 2, 1,
+ &mpc7_hdr[0], 2, 1, INIT_VLC_USE_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++){
+ 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_STATIC)){
+ av_log(avctx, AV_LOG_ERROR, "Cannot init QUANT VLC %i,%i\n",i,j);
+ return -1;
+ }
+ }
+ }
+ vlc_inited = 1;
+ return 0;
+}
+
+// XXX replace with something better
+static int av_always_inline mpc_rnd(MPCContext *c)
+{
+ c->rnd = c->rnd * 27 + 17;
+ return c->rnd;
+}
+
+/**
+ * Process decoded Musepack data and produce PCM
+ * @todo make it available for MPC8 and MPC6
+ */
+static void mpc_synth(MPCContext *c, int16_t *out)
+{
+ int dither_state = 0;
+ int i, ch;
+ OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
+
+ for(ch = 0; ch < 2; ch++){
+ samples_ptr = samples + ch;
+ for(i = 0; i < SAMPLES_PER_BAND; i++) {
+ ff_mpa_synth_filter(c->synth_buf[ch], &(c->synth_buf_offset[ch]),
+ mpa_window, &dither_state,
+ samples_ptr, 2,
+ c->sb_samples[ch][i]);
+ samples_ptr += 64;
+ }
+ }
+ for(i = 0; i < MPC_FRAME_SIZE*2; i++)
+ *out++=samples[i];
+}
+
+/**
+ * Fill samples for given subband
+ */
+static void inline idx_to_quant(MPCContext *c, GetBitContext *gb, int idx, int *dst)
+{
+ int i, i1, t;
+ switch(idx){
+ case -1:
+ for(i = 0; i < SAMPLES_PER_BAND; i++){
+ t = mpc_rnd(c);
+ *dst++ = ((t>>24)& 0xFF) + ((t>>16) & 0xFF) + ((t>>8) & 0xFF) + (t & 0xFF) - 510;
+ }
+ case 1:
+ i1 = get_bits1(gb);
+ for(i = 0; i < SAMPLES_PER_BAND/3; i++){
+ t = get_vlc2(gb, quant_vlc[0][i1].table, 9, 2);
+ *dst++ = mpc_idx30[t];
+ *dst++ = mpc_idx31[t];
+ *dst++ = mpc_idx32[t];
+ }
+ break;
+ case 2:
+ i1 = get_bits1(gb);
+ for(i = 0; i < SAMPLES_PER_BAND/2; i++){
+ t = get_vlc2(gb, quant_vlc[1][i1].table, 9, 2);
+ *dst++ = mpc_idx50[t];
+ *dst++ = mpc_idx51[t];
+ }
+ break;
+ case 3: case 4: case 5: case 6: case 7:
+ i1 = get_bits1(gb);
+ for(i = 0; i < SAMPLES_PER_BAND; i++)
+ *dst++ = get_vlc2(gb, 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++ = get_bits(gb, idx - 1) - t;
+ break;
+ default: // case 0 and -2..-17
+ return;
+ }
+}
+
+static int mpc7_decode_frame(AVCodecContext * avctx,
+ void *data, int *data_size,
+ uint8_t * buf, int buf_size)
+{
+ MPCContext *c = avctx->priv_data;
+ GetBitContext gb;
+ uint8_t *bits;
+ int i, j, ch, t;
+ int mb = -1;
+ Band bands[BANDS];
+ int Q[2][MPC_FRAME_SIZE];
+ int off;
+ float mul;
+ int bits_used, bits_avail;
+
+ memset(bands, 0, sizeof(bands));
+ if(buf_size <= 4){
+ av_log(avctx, AV_LOG_ERROR, "Too small buffer passed (%i bytes)\n", buf_size);
+ }
+
+ bits = av_malloc((buf_size - 1) & ~3);
+ c->dsp.bswap_buf(bits, buf + 4, (buf_size - 4) >> 2);
+ init_get_bits(&gb, bits, (buf_size - 4)* 8);
+ skip_bits(&gb, buf[0]);
+
+ /* read subband indexes */
+ for(i = 0; i <= c->bands; i++){
+ for(ch = 0; ch < 2; ch++){
+ if(i) t = get_vlc2(&gb, hdr_vlc.table, MPC7_HDR_BITS, 1) - 5;
+ if(!i || (t == 4)) bands[i].res[ch] = get_bits(&gb, 4);
+ else bands[i].res[ch] = bands[i-1].res[ch] + t;
+ }
+
+ if(bands[i].res[0] || bands[i].res[1]){
+ mb = i;
+ if(c->MSS) bands[i].msf = get_bits1(&gb);
+ }
+ }
+ /* 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] = get_vlc2(&gb, 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];
+ t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
+ bands[i].scf_idx[ch][0] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][2] + t);
+ switch(bands[i].scfi[ch]){
+ case 0:
+ t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
+ bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t);
+ t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
+ bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t);
+ break;
+ case 1:
+ t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
+ bands[i].scf_idx[ch][1] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][0] + t);
+ 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];
+ t = get_vlc2(&gb, dscf_vlc.table, MPC7_DSCF_BITS, 1) - 7;
+ bands[i].scf_idx[ch][2] = (t == 8) ? get_bits(&gb, 6) : (bands[i].scf_idx[ch][1] + t);
+ 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(Q, 0, sizeof(Q));
+ off = 0;
+ for(i = 0; i < BANDS; i++, off += SAMPLES_PER_BAND)
+ for(ch = 0; ch < 2; ch++)
+ idx_to_quant(c, &gb, bands[i].res[ch], Q[ch] + off);
+ /* dequantize */
+ memset(c->sb_samples, 0, sizeof(c->sb_samples));
+ off = 0;
+ for(i = 0; i <= mb; i++, off += SAMPLES_PER_BAND){
+ for(ch = 0; ch < 2; ch++){
+ if(bands[i].res[ch]){
+ j = 0;
+ mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][0]];
+ for(; j < 12; j++)
+ c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
+ mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][1]];
+ for(; j < 24; j++)
+ c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
+ mul = mpc_CC[bands[i].res[ch]] * mpc7_SCF[bands[i].scf_idx[ch][2]];
+ for(; j < 36; j++)
+ c->sb_samples[ch][j][i] = mul * Q[ch][j + off];
+ }
+ }
+ if(bands[i].msf){
+ int t1, t2;
+ for(j = 0; j < SAMPLES_PER_BAND; j++){
+ t1 = c->sb_samples[0][j][i];
+ t2 = c->sb_samples[1][j][i];
+ c->sb_samples[0][j][i] = t1 + t2;
+ c->sb_samples[1][j][i] = t1 - t2;
+ }
+ }
+ }
+
+ mpc_synth(c, data);
+
+ av_free(bits);
+
+ bits_used = get_bits_count(&gb);
+ bits_avail = (buf_size - 4) * 8;
+ if(!buf[1] && ((bits_avail < bits_used) || (bits_used + 32 <= bits_avail))){
+ av_log(NULL,0, "Error decoding frame: used %i of %i bits\n", bits_used, bits_avail);
+ return -1;
+ }
+ *data_size = (buf[1] ? c->lastframelen : MPC_FRAME_SIZE) * 4;
+
+ return buf_size;
+}
+
+
+AVCodec mpc7_decoder = {
+ "mpc sv7",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_MUSEPACK7,
+ sizeof(MPCContext),
+ mpc7_decode_init,
+ NULL,
+ NULL,
+ mpc7_decode_frame,
+};