* Musepack SV8 decoder
* Copyright (c) 2007 Konstantin Shishkov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file mpc8.c Musepack SV8 decoder
+ * @file
* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
* divided into 32 subbands.
*/
-#include "libavutil/random.h"
+#include "libavutil/lfg.h"
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "dsputil.h"
-
-#if CONFIG_MPEGAUDIO_HP
-#define USE_HIGHPRECISION
-#endif
-#include "mpegaudio.h"
+#include "mpegaudiodsp.h"
+#include "libavutil/audioconvert.h"
#include "mpc.h"
-#include "mpcdata.h"
#include "mpc8data.h"
#include "mpc8huff.h"
static inline int mpc8_dec_base(GetBitContext *gb, int k, int n)
{
- int code = get_bits(gb, mpc8_cnk_len[k-1][n-1] - 1);
+ int len = mpc8_cnk_len[k-1][n-1] - 1;
+ int code = len ? get_bits_long(gb, len) : 0;
if (code >= mpc8_cnk_lost[k-1][n-1])
code = ((code << 1) | get_bits1(gb)) - mpc8_cnk_lost[k-1][n-1];
return mask;
}
+static const uint16_t vlc_offsets[13] = {
+ 0, 640, 1184, 1748, 2298, 2426, 2554, 3066, 3578, 4106, 4618, 5196, 5708
+};
+
static av_cold int mpc8_decode_init(AVCodecContext * avctx)
{
int i;
MPCContext *c = avctx->priv_data;
GetBitContext gb;
static int vlc_initialized = 0;
+ int channels;
+
+ static VLC_TYPE band_table[542][2];
+ static VLC_TYPE q1_table[520][2];
+ static VLC_TYPE q9up_table[524][2];
+ static VLC_TYPE scfi0_table[1 << MPC8_SCFI0_BITS][2];
+ static VLC_TYPE scfi1_table[1 << MPC8_SCFI1_BITS][2];
+ static VLC_TYPE dscf0_table[560][2];
+ static VLC_TYPE dscf1_table[598][2];
+ static VLC_TYPE q3_0_table[512][2];
+ static VLC_TYPE q3_1_table[516][2];
+ static VLC_TYPE codes_table[5708][2];
if(avctx->extradata_size < 2){
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_init_random(0xDEADBEEF, &c->rnd);
- dsputil_init(&c->dsp, avctx);
+ av_lfg_init(&c->rnd, 0xDEADBEEF);
+ ff_dsputil_init(&c->dsp, avctx);
+ ff_mpadsp_init(&c->mpadsp);
ff_mpc_init();
skip_bits(&gb, 3);//sample rate
c->maxbands = get_bits(&gb, 5) + 1;
- skip_bits(&gb, 4);//channels
+ channels = get_bits(&gb, 4) + 1;
+ if (channels > 2) {
+ av_log_missing_feature(avctx, "Multichannel MPC SV8", 1);
+ return AVERROR_PATCHWELCOME;
+ }
c->MSS = get_bits1(&gb);
c->frames = 1 << (get_bits(&gb, 3) * 2);
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
+
if(vlc_initialized) return 0;
av_log(avctx, AV_LOG_DEBUG, "Initing VLC\n");
+ band_vlc.table = band_table;
+ band_vlc.table_allocated = 542;
init_vlc(&band_vlc, MPC8_BANDS_BITS, MPC8_BANDS_SIZE,
mpc8_bands_bits, 1, 1,
- mpc8_bands_codes, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_bands_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
+ q1_vlc.table = q1_table;
+ q1_vlc.table_allocated = 520;
init_vlc(&q1_vlc, MPC8_Q1_BITS, MPC8_Q1_SIZE,
mpc8_q1_bits, 1, 1,
- mpc8_q1_codes, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_q1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
+ q9up_vlc.table = q9up_table;
+ q9up_vlc.table_allocated = 524;
init_vlc(&q9up_vlc, MPC8_Q9UP_BITS, MPC8_Q9UP_SIZE,
mpc8_q9up_bits, 1, 1,
- mpc8_q9up_codes, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_q9up_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
+ scfi_vlc[0].table = scfi0_table;
+ scfi_vlc[0].table_allocated = 1 << MPC8_SCFI0_BITS;
init_vlc(&scfi_vlc[0], MPC8_SCFI0_BITS, MPC8_SCFI0_SIZE,
mpc8_scfi0_bits, 1, 1,
- mpc8_scfi0_codes, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_scfi0_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
+ scfi_vlc[1].table = scfi1_table;
+ scfi_vlc[1].table_allocated = 1 << MPC8_SCFI1_BITS;
init_vlc(&scfi_vlc[1], MPC8_SCFI1_BITS, MPC8_SCFI1_SIZE,
mpc8_scfi1_bits, 1, 1,
- mpc8_scfi1_codes, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_scfi1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
+ dscf_vlc[0].table = dscf0_table;
+ dscf_vlc[0].table_allocated = 560;
init_vlc(&dscf_vlc[0], MPC8_DSCF0_BITS, MPC8_DSCF0_SIZE,
mpc8_dscf0_bits, 1, 1,
- mpc8_dscf0_codes, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_dscf0_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
+ dscf_vlc[1].table = dscf1_table;
+ dscf_vlc[1].table_allocated = 598;
init_vlc(&dscf_vlc[1], MPC8_DSCF1_BITS, MPC8_DSCF1_SIZE,
mpc8_dscf1_bits, 1, 1,
- mpc8_dscf1_codes, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_dscf1_codes, 1, 1, INIT_VLC_USE_NEW_STATIC);
- init_vlc_sparse(&q3_vlc[0], MPC8_Q3_BITS, MPC8_Q3_SIZE,
+ q3_vlc[0].table = q3_0_table;
+ q3_vlc[0].table_allocated = 512;
+ ff_init_vlc_sparse(&q3_vlc[0], MPC8_Q3_BITS, MPC8_Q3_SIZE,
mpc8_q3_bits, 1, 1,
mpc8_q3_codes, 1, 1,
- mpc8_q3_syms, 1, 1, INIT_VLC_USE_STATIC);
- init_vlc_sparse(&q3_vlc[1], MPC8_Q4_BITS, MPC8_Q4_SIZE,
+ mpc8_q3_syms, 1, 1, INIT_VLC_USE_NEW_STATIC);
+ q3_vlc[1].table = q3_1_table;
+ q3_vlc[1].table_allocated = 516;
+ ff_init_vlc_sparse(&q3_vlc[1], MPC8_Q4_BITS, MPC8_Q4_SIZE,
mpc8_q4_bits, 1, 1,
mpc8_q4_codes, 1, 1,
- mpc8_q4_syms, 1, 1, INIT_VLC_USE_STATIC);
+ mpc8_q4_syms, 1, 1, INIT_VLC_USE_NEW_STATIC);
for(i = 0; i < 2; i++){
+ res_vlc[i].table = &codes_table[vlc_offsets[0+i]];
+ res_vlc[i].table_allocated = vlc_offsets[1+i] - vlc_offsets[0+i];
init_vlc(&res_vlc[i], MPC8_RES_BITS, MPC8_RES_SIZE,
&mpc8_res_bits[i], 1, 1,
- &mpc8_res_codes[i], 1, 1, INIT_VLC_USE_STATIC);
+ &mpc8_res_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
+ q2_vlc[i].table = &codes_table[vlc_offsets[2+i]];
+ q2_vlc[i].table_allocated = vlc_offsets[3+i] - vlc_offsets[2+i];
init_vlc(&q2_vlc[i], MPC8_Q2_BITS, MPC8_Q2_SIZE,
&mpc8_q2_bits[i], 1, 1,
- &mpc8_q2_codes[i], 1, 1, INIT_VLC_USE_STATIC);
+ &mpc8_q2_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
+ quant_vlc[0][i].table = &codes_table[vlc_offsets[4+i]];
+ quant_vlc[0][i].table_allocated = vlc_offsets[5+i] - vlc_offsets[4+i];
init_vlc(&quant_vlc[0][i], MPC8_Q5_BITS, MPC8_Q5_SIZE,
&mpc8_q5_bits[i], 1, 1,
- &mpc8_q5_codes[i], 1, 1, INIT_VLC_USE_STATIC);
+ &mpc8_q5_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
+ quant_vlc[1][i].table = &codes_table[vlc_offsets[6+i]];
+ quant_vlc[1][i].table_allocated = vlc_offsets[7+i] - vlc_offsets[6+i];
init_vlc(&quant_vlc[1][i], MPC8_Q6_BITS, MPC8_Q6_SIZE,
&mpc8_q6_bits[i], 1, 1,
- &mpc8_q6_codes[i], 1, 1, INIT_VLC_USE_STATIC);
+ &mpc8_q6_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
+ quant_vlc[2][i].table = &codes_table[vlc_offsets[8+i]];
+ quant_vlc[2][i].table_allocated = vlc_offsets[9+i] - vlc_offsets[8+i];
init_vlc(&quant_vlc[2][i], MPC8_Q7_BITS, MPC8_Q7_SIZE,
&mpc8_q7_bits[i], 1, 1,
- &mpc8_q7_codes[i], 1, 1, INIT_VLC_USE_STATIC);
+ &mpc8_q7_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
+ quant_vlc[3][i].table = &codes_table[vlc_offsets[10+i]];
+ quant_vlc[3][i].table_allocated = vlc_offsets[11+i] - vlc_offsets[10+i];
init_vlc(&quant_vlc[3][i], MPC8_Q8_BITS, MPC8_Q8_SIZE,
&mpc8_q8_bits[i], 1, 1,
- &mpc8_q8_codes[i], 1, 1, INIT_VLC_USE_STATIC);
+ &mpc8_q8_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
}
vlc_initialized = 1;
- avctx->sample_fmt = SAMPLE_FMT_S16;
- avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
+
+ avcodec_get_frame_defaults(&c->frame);
+ avctx->coded_frame = &c->frame;
+
return 0;
}
-static int mpc8_decode_frame(AVCodecContext * avctx,
- void *data, int *data_size,
- const uint8_t * buf, int buf_size)
+static int mpc8_decode_frame(AVCodecContext * avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
MPCContext *c = avctx->priv_data;
GetBitContext gb2, *gb = &gb2;
int i, j, k, ch, cnt, res, t;
int maxband, keyframe;
int last[2];
+ /* get output buffer */
+ c->frame.nb_samples = MPC_FRAME_SIZE;
+ if ((res = avctx->get_buffer(avctx, &c->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return res;
+ }
+
keyframe = c->cur_frame == 0;
if(keyframe){
maxband = c->last_max_band + get_vlc2(gb, band_vlc.table, MPC8_BANDS_BITS, 2);
if(maxband > 32) maxband -= 33;
}
+ if(maxband > c->maxbands + 1)
+ return AVERROR_INVALIDDATA;
c->last_max_band = maxband;
/* read subband indexes */
switch(res){
case -1:
for(j = 0; j < SAMPLES_PER_BAND; j++)
- c->Q[ch][off + j] = (av_random(&c->rnd) & 0x3FC) - 510;
+ c->Q[ch][off + j] = (av_lfg_get(&c->rnd) & 0x3FC) - 510;
break;
case 0:
break;
}
}
- ff_mpc_dequantize_and_synth(c, maxband, data);
+ ff_mpc_dequantize_and_synth(c, maxband - 1,
+ (int16_t **)c->frame.extended_data,
+ avctx->channels);
c->cur_frame++;
c->last_bits_used = get_bits_count(gb);
+ if(get_bits_left(gb) < 8) // we have only padding left
+ c->last_bits_used = buf_size << 3;
if(c->cur_frame >= c->frames)
c->cur_frame = 0;
- *data_size = MPC_FRAME_SIZE * 4;
+
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = c->frame;
return c->cur_frame ? c->last_bits_used >> 3 : buf_size;
}
-AVCodec mpc8_decoder = {
- "mpc8",
- CODEC_TYPE_AUDIO,
- CODEC_ID_MUSEPACK8,
- sizeof(MPCContext),
- mpc8_decode_init,
- NULL,
- NULL,
- mpc8_decode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"),
+static av_cold void mpc8_decode_flush(AVCodecContext *avctx)
+{
+ MPCContext *c = avctx->priv_data;
+ c->cur_frame = 0;
+}
+
+AVCodec ff_mpc8_decoder = {
+ .name = "mpc8",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_MUSEPACK8,
+ .priv_data_size = sizeof(MPCContext),
+ .init = mpc8_decode_init,
+ .decode = mpc8_decode_frame,
+ .flush = mpc8_decode_flush,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("Musepack SV8"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff