* 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
* especially on files encoded with 16 or 8 subbands.
*/
#include <stddef.h>
#include <stdio.h>
-#define ALT_BITSTREAM_READER_LE
+#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
+#include "rdft.h"
+#include "mpegaudiodsp.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) { \
#define SAMPLES_NEEDED_2(why) \
av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
+#define QDM2_MAX_FRAME_SIZE 512
typedef int8_t sb_int8_array[2][30][64];
} FFTCoefficient;
typedef struct {
- DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]);
+ DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
* QDM2 decoder context
*/
typedef struct {
+ AVFrame frame;
+
/// Parameters from codec header, do not change during playback
int nb_channels; ///< number of channels
int channels; ///< number of channels
/// I/O data
const uint8_t *compressed_data;
int compressed_size;
- float output_buffer[1024];
+ float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
/// Synthesis filter
- DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
+ MPADSPContext mpadsp;
+ DECLARE_ALIGNED(32, float, 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(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+ DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
/// 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 DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
-
-
-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,
};
if (!vlcs_initialized) {
- vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
- vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
- init_vlc (&vlc_tab_level, 8, 24,
- vlc_tab_level_huffbits, 1, 1,
- vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
- vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
- init_vlc (&vlc_tab_diff, 8, 37,
- vlc_tab_diff_huffbits, 1, 1,
- vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
- vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
- init_vlc (&vlc_tab_run, 5, 6,
- vlc_tab_run_huffbits, 1, 1,
- vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
- fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
- init_vlc (&fft_level_exp_alt_vlc, 8, 28,
- fft_level_exp_alt_huffbits, 1, 1,
- fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
-
- fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
- fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
- init_vlc (&fft_level_exp_vlc, 8, 20,
- fft_level_exp_huffbits, 1, 1,
- fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
- fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
- init_vlc (&fft_stereo_exp_vlc, 6, 7,
- fft_stereo_exp_huffbits, 1, 1,
- fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
- fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
- init_vlc (&fft_stereo_phase_vlc, 6, 9,
- fft_stereo_phase_huffbits, 1, 1,
- fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
- vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
- init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
- vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
- vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
- vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
- init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
- vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
- vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
- vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
- init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
- vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
- vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
- vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
- init_vlc (&vlc_tab_type30, 6, 9,
- vlc_tab_type30_huffbits, 1, 1,
- vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
- vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
- init_vlc (&vlc_tab_type34, 5, 10,
- vlc_tab_type34_huffbits, 1, 1,
- vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
- vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
- init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
- vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
- vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
- init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
- vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
- vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
- init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
- vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
- vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
- init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
- vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
-
- vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
- vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
- init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
- vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
- vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+ vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
+ vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
+ init_vlc (&vlc_tab_level, 8, 24,
+ vlc_tab_level_huffbits, 1, 1,
+ vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
+ vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
+ init_vlc (&vlc_tab_diff, 8, 37,
+ vlc_tab_diff_huffbits, 1, 1,
+ vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
+ vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
+ init_vlc (&vlc_tab_run, 5, 6,
+ vlc_tab_run_huffbits, 1, 1,
+ vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
+ fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
+ init_vlc (&fft_level_exp_alt_vlc, 8, 28,
+ fft_level_exp_alt_huffbits, 1, 1,
+ fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+
+ fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
+ fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
+ init_vlc (&fft_level_exp_vlc, 8, 20,
+ fft_level_exp_huffbits, 1, 1,
+ fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
+ fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
+ init_vlc (&fft_stereo_exp_vlc, 6, 7,
+ fft_stereo_exp_huffbits, 1, 1,
+ fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
+ fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
+ init_vlc (&fft_stereo_phase_vlc, 6, 9,
+ fft_stereo_phase_huffbits, 1, 1,
+ fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
+ vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
+ init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
+ vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
+ vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
+ vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
+ init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
+ vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
+ vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
+ vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
+ init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
+ vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
+ vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
+ vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
+ init_vlc (&vlc_tab_type30, 6, 9,
+ vlc_tab_type30_huffbits, 1, 1,
+ vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
+ vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
+ init_vlc (&vlc_tab_type34, 5, 10,
+ vlc_tab_type34_huffbits, 1, 1,
+ vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
+ vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
+ init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
+ vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
+ vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
+ vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
+ init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
+ vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
+ vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
+ vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
+ init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
+ vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
+ vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
+ vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
+ init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
+ vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
+ vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
+
+ vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
+ vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
+ init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
+ vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
+ vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
vlcs_initialized=1;
}
}
-
-/* for floating point to fixed point conversion */
-static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
-
-
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
{
int value;
/**
- * 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
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 64; j++) {
- q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
- q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
+ q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
+ q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
}
}
for (chs = 0; chs < q->nb_channels; chs++)
for (k = 0; k < run; k++)
if ((j + k) < 128)
- q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
+ q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
} else {
for (k = 0; k < run; k++)
if ((j + k) < 128)
- q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
+ q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
}
j += run;
* 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;
local_int_14 = (offset >> local_int_8);
+ if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
+ return;
if (q->nb_channels > 1) {
channel = get_bits1(gb);
int i;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
- ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
+ q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
/* add samples to output buffer */
for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
*/
static void qdm2_synthesis_filter (QDM2Context *q, int index)
{
- OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
/* copy sb_samples */
q->sb_samples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < q->nb_channels; ch++) {
- OUT_INT *samples_ptr = samples + ch;
+ float *samples_ptr = q->samples + ch;
for (i = 0; i < 8; i++) {
- ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
- mpa_window, &dither_state,
+ ff_mpa_synth_filter_float(&q->mpadsp,
+ q->synth_buf[ch], &(q->synth_buf_offset[ch]),
+ ff_mpa_synth_window_float, &dither_state,
samples_ptr, q->nb_channels,
q->sb_samples[ch][(8 * index) + i]);
samples_ptr += 32 * q->nb_channels;
for (ch = 0; ch < q->channels; ch++)
for (i = 0; i < q->frame_size; i++)
- q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
+ q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
}
initialized = 1;
qdm2_init_vlc();
- ff_mpa_synth_init(mpa_window);
+ ff_mpa_synth_init_float(ff_mpa_synth_window_float);
softclip_table_init();
rnd_table_init();
init_noise_samples();
avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
extradata += 4;
+ if (s->channels > MPA_MAX_CHANNELS)
+ return AVERROR_INVALIDDATA;
avctx->sample_rate = AV_RB32(extradata);
extradata += 4;
// something like max decodable tones
s->group_order = av_log2(s->group_size) + 1;
s->frame_size = s->group_size / 16; // 16 iterations per super block
+ if (s->frame_size > QDM2_MAX_FRAME_SIZE)
+ return AVERROR_INVALIDDATA;
s->sub_sampling = s->fft_order - 7;
s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
return -1;
}
- ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
+ ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
+ ff_mpadsp_init(&s->mpadsp);
qdm2_init(s);
- avctx->sample_fmt = SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
// 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;
}
-static int qdm2_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
+ int16_t *out;
+ int i, ret;
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);
+ /* get output buffer */
+ s->frame.nb_samples = 16 * s->frame_size;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ out = (int16_t *)s->frame.data[0];
- // 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;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
+ 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,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
};
projects / ffmpeg / blobdiff