* Copyright (c) 2005 Alex Beregszaszi
* Copyright (c) 2005 Roberto Togni
*
- * 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 libavcodec/qdm2.c
+ * @file
* QDM2 decoder
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
* The decoder is not perfect yet, there are still some distortions
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
+#include "fft.h"
#include "mpegaudio.h"
#include "qdm2data.h"
+#include "qdm2_tablegen.h"
#undef NDEBUG
#include <assert.h>
-#define SOFTCLIP_THRESHOLD 27600
-#define HARDCLIP_THRESHOLD 35716
-
-
#define QDM2_LIST_ADD(list, size, packet) \
do { \
if (size > 0) { \
} FFTCoefficient;
typedef struct {
- DECLARE_ALIGNED_16(QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
+ DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
float output_buffer[1024];
/// Synthesis filter
- DECLARE_ALIGNED_16(MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
+ 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][128][SBLIMIT];
+ DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
/// Mixed temporary data used in decoding
float tone_level[MPA_MAX_CHANNELS][30][64];
static VLC vlc_tab_type34;
static VLC vlc_tab_fft_tone_offset[5];
-static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
-static float noise_table[4096];
-static uint8_t random_dequant_index[256][5];
-static uint8_t random_dequant_type24[128][3];
-static float noise_samples[128];
-
-
-static av_cold void softclip_table_init(void) {
- int i;
- double dfl = SOFTCLIP_THRESHOLD - 32767;
- float delta = 1.0 / -dfl;
- for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
- softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
-}
-
-
-// random generated table
-static av_cold void rnd_table_init(void) {
- int i,j;
- uint32_t ldw,hdw;
- uint64_t tmp64_1;
- uint64_t random_seed = 0;
- float delta = 1.0 / 16384.0;
- for(i = 0; i < 4096 ;i++) {
- random_seed = random_seed * 214013 + 2531011;
- noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
- }
-
- for (i = 0; i < 256 ;i++) {
- random_seed = 81;
- ldw = i;
- for (j = 0; j < 5 ;j++) {
- random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
- ldw = (uint32_t)ldw % (uint32_t)random_seed;
- tmp64_1 = (random_seed * 0x55555556);
- hdw = (uint32_t)(tmp64_1 >> 32);
- random_seed = (uint64_t)(hdw + (ldw >> 31));
- }
- }
- for (i = 0; i < 128 ;i++) {
- random_seed = 25;
- ldw = i;
- for (j = 0; j < 3 ;j++) {
- random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
- ldw = (uint32_t)ldw % (uint32_t)random_seed;
- tmp64_1 = (random_seed * 0x66666667);
- hdw = (uint32_t)(tmp64_1 >> 33);
- random_seed = hdw + (ldw >> 31);
- }
- }
-}
-
-
-static av_cold void init_noise_samples(void) {
- int i;
- int random_seed = 0;
- float delta = 1.0 / 16384.0;
- for (i = 0; i < 128;i++) {
- random_seed = random_seed * 214013 + 2531011;
- noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
- }
-}
-
static const uint16_t qdm2_vlc_offs[] = {
0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
};
/**
- * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
+ * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
*
* @param gb bitreader context
* @param sub_packet packet under analysis
/**
- * Replaces 8 elements with their average value.
+ * Replace 8 elements with their average value.
* Called by qdm2_decode_superblock before starting subblock decoding.
*
* @param q context
* This is similar to process_subpacket_9, but for a single channel and for element [0]
* same VLC tables as process_subpacket_9 are used.
*
- * @param q context
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
* @param gb bitreader context
* @param length packet length in bits
init_get_bits(&gb, header.data, header.size*8);
if (header.type == 2 || header.type == 4 || header.type == 5) {
- int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
+ int csum = 257 * get_bits(&gb, 8);
+ csum += 2 * get_bits(&gb, 8);
csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
return -1;
}
- ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
+ ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
qdm2_init(s);
- avctx->sample_fmt = SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
// dump_context(s);
return 0;
}
-static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
+static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
{
int ch, i;
const int frame_size = (q->frame_size * q->channels);
if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
SAMPLES_NEEDED_2("has errors, and C list is not empty")
- return;
+ return -1;
}
}
out[i] = value;
}
+
+ return 0;
}
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
+ int16_t *out = data;
+ int i;
if(!buf)
return 0;
if(buf_size < s->checksum_size)
return -1;
- *data_size = s->channels * s->frame_size * sizeof(int16_t);
-
av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
buf_size, buf, s->checksum_size, data, *data_size);
- qdm2_decode(s, buf, data);
-
- // reading only when next superblock found
- if (s->sub_packet == 0) {
- return s->checksum_size;
+ for (i = 0; i < 16; i++) {
+ if (qdm2_decode(s, buf, out) < 0)
+ return -1;
+ out += s->channels * s->frame_size;
}
- return 0;
+ *data_size = (uint8_t*)out - (uint8_t*)data;
+
+ return s->checksum_size;
}
-AVCodec qdm2_decoder =
+AVCodec ff_qdm2_decoder =
{
.name = "qdm2",
- .type = CODEC_TYPE_AUDIO,
+ .type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_QDM2,
.priv_data_size = sizeof(QDM2Context),
.init = qdm2_decode_init,
projects / ffmpeg / blobdiff