* Copyright (c) 2005 Alex Beregszaszi
* Copyright (c) 2005 Roberto Togni
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library 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 this library; 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 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 "bitstream.h"
+#include "get_bits.h"
#include "dsputil.h"
-
-#ifdef CONFIG_MPEGAUDIO_HP
-#define USE_HIGHPRECISION
-#endif
-
+#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 SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
-#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
-
#define SAMPLES_NEEDED \
av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
#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];
/**
* A node in the subpacket list
*/
-typedef struct _QDM2SubPNode {
+typedef struct QDM2SubPNode {
QDM2SubPacket *packet; ///< packet
- struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
+ struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
} QDM2SubPNode;
+typedef struct {
+ float re;
+ float im;
+} QDM2Complex;
+
typedef struct {
float level;
- float *samples_im;
- float *samples_re;
- float *table;
+ QDM2Complex *complex;
+ const float *table;
int phase;
int phase_shift;
int duration;
} FFTCoefficient;
typedef struct {
- float re;
- float im;
-} QDM2Complex;
-
-typedef struct {
- QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
- float samples_im[MPA_MAX_CHANNELS][256];
- float samples_re[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
/// Parameters built from header parameters, do not change during playback
int group_order; ///< order of frame group
int fft_order; ///< order of FFT (actually fftorder+1)
- int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
int frame_size; ///< size of data frame
int frequency_range;
int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
int fft_coefs_min_index[5];
int fft_coefs_max_index[5];
int fft_level_exp[6];
- FFTContext fft_ctx;
- FFTComplex exptab[128];
+ RDFTContext rdft_ctx;
QDM2FFT fft;
/// I/O data
- uint8_t *compressed_data;
+ const uint8_t *compressed_data;
int compressed_size;
- float output_buffer[1024];
+ float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
/// Synthesis filter
- MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
+ MPADSPContext mpadsp;
+ DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
int synth_buf_offset[MPA_MAX_CHANNELS];
- int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
+ 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];
} QDM2Context;
-static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
-
static VLC vlc_tab_level;
static VLC vlc_tab_diff;
static VLC vlc_tab_run;
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 MPA_INT mpa_window[512] __attribute__((aligned(16)));
-
-
-static 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 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 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,
+};
-static void qdm2_init_vlc(void)
+static av_cold void qdm2_init_vlc(void)
{
- init_vlc (&vlc_tab_level, 8, 24,
- vlc_tab_level_huffbits, 1, 1,
- vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- init_vlc (&vlc_tab_diff, 8, 37,
- vlc_tab_diff_huffbits, 1, 1,
- vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- init_vlc (&vlc_tab_run, 5, 6,
- vlc_tab_run_huffbits, 1, 1,
- vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- init_vlc (&fft_level_exp_vlc, 8, 20,
- fft_level_exp_huffbits, 1, 1,
- fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- init_vlc (&fft_stereo_exp_vlc, 6, 7,
- fft_stereo_exp_huffbits, 1, 1,
- fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- init_vlc (&fft_stereo_phase_vlc, 6, 9,
- fft_stereo_phase_huffbits, 1, 1,
- fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- init_vlc (&vlc_tab_type30, 6, 9,
- vlc_tab_type30_huffbits, 1, 1,
- vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- init_vlc (&vlc_tab_type34, 5, 10,
- vlc_tab_type34_huffbits, 1, 1,
- vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
-
- 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_STATIC | INIT_VLC_LE);
+ static int vlcs_initialized = 0;
+ static VLC_TYPE qdm2_table[3838][2];
+
+ 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);
+
+ vlcs_initialized=1;
+ }
}
-
-/* for floating point to fixed point conversion */
-static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
-
-
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
{
int value;
*
* @return 0 if checksum is OK
*/
-static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
+static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
int i;
for (i=0; i < length; i++)
/**
- * 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];
}
}
int j,k;
int ch;
int run, case_val;
- int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
+ static const int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
for (ch = 0; ch < channels; ch++) {
for (j = 0; j < 64; ) {
run = 1;
case_val = 8;
} else {
- switch (switchtable[coding_method[ch][sb][j]]) {
+ switch (switchtable[coding_method[ch][sb][j]-8]) {
case 0: run = 10; case_val = 10; break;
case 1: run = 1; case_val = 16; break;
case 2: run = 5; case_val = 24; break;
SAMPLES_NEEDED
for (ch = 0; ch < nb_channels; ch++)
for (sb = 0; sb < 30; sb++) {
- for (j = 1; j < 64; j++) {
+ for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
add1 = tone_level_idx[ch][sb][j] - 10;
if (add1 < 0)
add1 = 0;
for (sb = 0; sb < 30; sb++)
for (j = 0; j < 64; j++)
acc += tone_level_idx_temp[ch][sb][j];
- if (acc)
- tmp = c * 256 / (acc & 0xffff);
+
multres = 0x66666667 * (acc * 10);
esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
for (ch = 0; ch < nb_channels; ch++)
else if (sb >= 24)
joined_stereo = 1;
else
- joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
+ joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
if (joined_stereo) {
- if (BITS_LEFT(length,gb) >= 16)
+ if (get_bits_left(gb) >= 16)
for (j = 0; j < 16; j++)
sign_bits[j] = get_bits1 (gb);
}
for (ch = 0; ch < channels; ch++) {
- zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
+ zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
type34_predictor = 0.0;
type34_first = 1;
for (j = 0; j < 128; ) {
switch (q->coding_method[ch][sb][j / 2]) {
case 8:
- if (BITS_LEFT(length,gb) >= 10) {
+ if (get_bits_left(gb) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + 2 * k) >= 128)
break;
case 10:
- if (BITS_LEFT(length,gb) >= 1) {
+ if (get_bits_left(gb) >= 1) {
float f = 0.81;
if (get_bits1(gb))
break;
case 16:
- if (BITS_LEFT(length,gb) >= 10) {
+ if (get_bits_left(gb) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + k) >= 128)
break;
case 24:
- if (BITS_LEFT(length,gb) >= 7) {
+ if (get_bits_left(gb) >= 7) {
n = get_bits(gb, 7);
for (k = 0; k < 3; k++)
samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
break;
case 30:
- if (BITS_LEFT(length,gb) >= 4)
- samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
- else
+ if (get_bits_left(gb) >= 4) {
+ unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
+ if (index < FF_ARRAY_ELEMS(type30_dequant)) {
+ samples[0] = type30_dequant[index];
+ } else
+ samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
+ } else
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
run = 1;
break;
case 34:
- if (BITS_LEFT(length,gb) >= 7) {
+ if (get_bits_left(gb) >= 7) {
if (type34_first) {
type34_div = (float)(1 << get_bits(gb, 2));
samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
type34_predictor = samples[0];
type34_first = 0;
} else {
- samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
- type34_predictor = samples[0];
+ unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
+ if (index < FF_ARRAY_ELEMS(type34_delta)) {
+ samples[0] = type34_delta[index] / type34_div + type34_predictor;
+ type34_predictor = samples[0];
+ } else
+ samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
}
} else {
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
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
*/
-static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
+static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
{
int i, k, run, level, diff;
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
return;
level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
quantized_coeffs[0] = level;
for (i = 0; i < 7; ) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
*
* @param q context
* @param gb bitreader context
- * @param length packet length in bits
*/
-static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
+static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
{
int sb, j, k, n, ch;
for (ch = 0; ch < q->nb_channels; ch++) {
- init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
+ init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
- if (BITS_LEFT(length,gb) < 16) {
+ if (get_bits_left(gb) < 16) {
memset(q->quantized_coeffs[ch][0], 0, 8);
break;
}
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 1)
+ if (get_bits_left(gb) < 1)
break;
if (get_bits1(gb)) {
for (k=0; k < 8; k++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
}
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
if (sb > 19)
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (BITS_LEFT(length,gb) < 16)
+ if (get_bits_left(gb) < 16)
break;
q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
}
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bits
*/
-static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
-
- if (length != 0) {
- init_tone_level_dequantization(q, &gb, length);
+ if (node) {
+ init_get_bits(&gb, node->packet->data, node->packet->size * 8);
+ init_tone_level_dequantization(q, &gb);
fill_tone_level_array(q, 1);
} else {
fill_tone_level_array(q, 0);
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bit
*/
-static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
+ int length = 0;
+
+ if (node) {
+ length = node->packet->size * 8;
+ init_get_bits(&gb, node->packet->data, length);
+ }
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
if (length >= 32) {
int c = get_bits (&gb, 13);
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bits
*/
-static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
+static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
+ int length = 0;
+
+ if (node) {
+ length = node->packet->size * 8;
+ init_get_bits(&gb, node->packet->data, length);
+ }
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
}
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
if (nodes[1] != NULL)
- process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
+ process_subpacket_10(q, nodes[1]);
else
- process_subpacket_10(q, NULL, 0);
+ process_subpacket_10(q, NULL);
nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
- process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
+ process_subpacket_11(q, nodes[2]);
else
- process_subpacket_11(q, NULL, 0);
+ process_subpacket_11(q, NULL);
nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
- process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
+ process_subpacket_12(q, nodes[3]);
else
- process_subpacket_12(q, NULL, 0);
+ process_subpacket_12(q, NULL);
}
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);
process_synthesis_subpackets(q, q->sub_packet_list_D);
q->do_synth_filter = 1;
} else if (q->do_synth_filter) {
- process_subpacket_10(q, NULL, 0);
- process_subpacket_11(q, NULL, 0);
- process_subpacket_12(q, NULL, 0);
+ process_subpacket_10(q, NULL);
+ process_subpacket_11(q, NULL);
+ process_subpacket_12(q, NULL);
}
/* **************************************************************** */
}
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);
if (q->sub_packet_list_B[0].packet == NULL)
return;
- /* reset minimum indices for FFT coefficients */
+ /* reset minimum indexes for FFT coefficients */
q->fft_coefs_index = 0;
for (i=0; i < 5; i++)
q->fft_coefs_min_index[i] = -1;
/* process subpackets ordered by type, largest type first */
for (i = 0, max = 256; i < q->sub_packets_B; i++) {
- QDM2SubPacket *packet;
+ QDM2SubPacket *packet= NULL;
/* find subpacket with largest type less than max */
- for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
+ for (j = 0, min = 0; j < q->sub_packets_B; j++) {
value = q->sub_packet_list_B[j].packet->type;
if (value > min && value < max) {
min = value;
max = min;
/* check for errors (?) */
+ if (!packet)
+ return;
+
if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
return;
if (duration >= 0 && duration < 4)
qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
} else if (type == 31) {
- for (i=0; i < 4; i++)
- qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
+ for (j=0; j < 4; j++)
+ qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
} else if (type == 46) {
- for (i=0; i < 6; i++)
- q->fft_level_exp[i] = get_bits(&gb, 6);
- for (i=0; i < 4; i++)
- qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
+ for (j=0; j < 6; j++)
+ q->fft_level_exp[j] = get_bits(&gb, 6);
+ for (j=0; j < 4; j++)
+ qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
}
} // Loop on B packets
- /* calculate maximum indices for FFT coefficients */
+ /* calculate maximum indexes for FFT coefficients */
for (i = 0, j = -1; i < 5; i++)
if (q->fft_coefs_min_index[i] >= 0) {
if (j >= 0)
/* generate FFT coefficients for tone */
if (tone->duration >= 3 || tone->cutoff >= 3) {
- tone->samples_im[0] += c.im;
- tone->samples_re[0] += c.re;
- tone->samples_im[1] -= c.im;
- tone->samples_re[1] -= c.re;
+ tone->complex[0].im += c.im;
+ tone->complex[0].re += c.re;
+ tone->complex[1].im -= c.im;
+ tone->complex[1].re -= c.re;
} else {
f[1] = -tone->table[4];
f[0] = tone->table[3] - tone->table[0];
f[4] = tone->table[0] - tone->table[1];
f[5] = tone->table[2];
for (i = 0; i < 2; i++) {
- tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
- tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
+ tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
+ tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
}
for (i = 0; i < 4; i++) {
- tone->samples_re[i] += c.re * f[i+2];
- tone->samples_im[i] += c.im * f[i+2];
+ tone->complex[i].re += c.re * f[i+2];
+ tone->complex[i].im += c.im * f[i+2];
}
}
const double iscale = 0.25 * M_PI;
for (ch = 0; ch < q->channels; ch++) {
- memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
- memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
+ memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
}
c.re = level * cos(q->fft_coefs[i].phase * iscale);
c.im = level * sin(q->fft_coefs[i].phase * iscale);
- q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
- q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
- q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
- q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
+ q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
+ q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
+ q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
+ q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
}
/* generate existing FFT tones */
tone.cutoff = (offset >= 60) ? 3 : 2;
tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
- tone.samples_im = &q->fft.samples_im[ch][offset];
- tone.samples_re = &q->fft.samples_re[ch][offset];
- tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
+ tone.complex = &q->fft.complex[ch][offset];
+ tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
tone.duration = i;
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
{
- const int n = 1 << (q->fft_order - 1);
- const int n2 = n >> 1;
- const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
- float c, s, f0, f1, f2, f3;
- int i, j;
-
- /* prerotation (or something like that) */
- for (i=1; i < n2; i++) {
- j = (n - i);
- c = q->exptab[i].re;
- s = -q->exptab[i].im;
- f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
- f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
- f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
- f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
- q->fft.complex[i].re = s * f0 - c * f1 + f2;
- q->fft.complex[i].im = c * f0 + s * f1 + f3;
- q->fft.complex[j].re = -s * f0 + c * f1 + f2;
- q->fft.complex[j].im = c * f0 + s * f1 - f3;
- }
-
- q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0;
- q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0;
- q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0;
- q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
-
- ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
- ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
+ const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
+ float *out = q->output_buffer + channel;
+ int i;
+ q->fft.complex[channel][0].re *= 2.0f;
+ q->fft.complex[channel][0].im = 0.0f;
+ 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)[i];
+ for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
+ out[0] += q->fft.complex[channel][i].re * gain;
+ out[q->channels] += q->fft.complex[channel][i].im * gain;
+ out += 2 * q->channels;
+ }
}
*/
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];
}
*
* @param q context
*/
-static void qdm2_init(QDM2Context *q) {
- static int inited = 0;
+static av_cold void qdm2_init(QDM2Context *q) {
+ static int initialized = 0;
- if (inited != 0)
+ if (initialized != 0)
return;
- inited = 1;
+ 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();
}
-#if 0
-static void dump_context(QDM2Context *q)
-{
- int i;
-#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
- PRINT("compressed_data",q->compressed_data);
- PRINT("compressed_size",q->compressed_size);
- PRINT("frame_size",q->frame_size);
- PRINT("checksum_size",q->checksum_size);
- PRINT("channels",q->channels);
- PRINT("nb_channels",q->nb_channels);
- PRINT("fft_frame_size",q->fft_frame_size);
- PRINT("fft_size",q->fft_size);
- PRINT("sub_sampling",q->sub_sampling);
- PRINT("fft_order",q->fft_order);
- PRINT("group_order",q->group_order);
- PRINT("group_size",q->group_size);
- PRINT("sub_packet",q->sub_packet);
- PRINT("frequency_range",q->frequency_range);
- PRINT("has_errors",q->has_errors);
- PRINT("fft_tone_end",q->fft_tone_end);
- PRINT("fft_tone_start",q->fft_tone_start);
- PRINT("fft_coefs_index",q->fft_coefs_index);
- PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
- PRINT("cm_table_select",q->cm_table_select);
- PRINT("noise_idx",q->noise_idx);
-
- for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
- {
- FFTTone *t = &q->fft_tones[i];
-
- av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
- av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
-// PRINT(" level", t->level);
- PRINT(" phase", t->phase);
- PRINT(" phase_shift", t->phase_shift);
- PRINT(" duration", t->duration);
- PRINT(" samples_im", t->samples_im);
- PRINT(" samples_re", t->samples_re);
- PRINT(" table", t->table);
- }
-
-}
-#endif
-
-
/**
* Init parameters from codec extradata
*/
-static int qdm2_decode_init(AVCodecContext *avctx)
+static av_cold int qdm2_decode_init(AVCodecContext *avctx)
{
QDM2Context *s = avctx->priv_data;
uint8_t *extradata;
int extradata_size;
int tmp_val, tmp, size;
- int i;
- float alpha;
/* extradata parsing
extradata += 8;
extradata_size -= 8;
- size = BE_32(extradata);
+ size = AV_RB32(extradata);
if(size > extradata_size){
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
extradata += 4;
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
- if (BE_32(extradata) != MKBETAG('Q','D','C','A')) {
+ if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
return -1;
}
extradata += 8;
- avctx->channels = s->nb_channels = s->channels = BE_32(extradata);
+ avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
extradata += 4;
+ if (s->channels > MPA_MAX_CHANNELS)
+ return AVERROR_INVALIDDATA;
- avctx->sample_rate = BE_32(extradata);
+ avctx->sample_rate = AV_RB32(extradata);
extradata += 4;
- avctx->bit_rate = BE_32(extradata);
+ avctx->bit_rate = AV_RB32(extradata);
extradata += 4;
- s->group_size = BE_32(extradata);
+ s->group_size = AV_RB32(extradata);
extradata += 4;
- s->fft_size = BE_32(extradata);
+ s->fft_size = AV_RB32(extradata);
extradata += 4;
- s->checksum_size = BE_32(extradata);
- extradata += 4;
+ s->checksum_size = AV_RB32(extradata);
+ if (s->checksum_size >= 1U << 28) {
+ av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
+ return AVERROR_INVALIDDATA;
+ }
s->fft_order = av_log2(s->fft_size) + 1;
- s->fft_frame_size = 2 * s->fft_size; // complex has two floats
// 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));
else
s->coeff_per_sb_select = 2;
- // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
+ // Fail on unknown fft order
if ((s->fft_order < 7) || (s->fft_order > 9)) {
av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
return -1;
}
- ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
-
- for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
- alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
- s->exptab[i].re = cos(alpha);
- s->exptab[i].im = sin(alpha);
- }
+ ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
+ ff_mpadsp_init(&s->mpadsp);
qdm2_init(s);
-// dump_context(s);
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
-static int qdm2_decode_close(AVCodecContext *avctx)
+static av_cold int qdm2_decode_close(AVCodecContext *avctx)
{
QDM2Context *s = avctx->priv_data;
- ff_fft_end(&s->fft_ctx);
+ ff_rdft_end(&s->rdft_ctx);
return 0;
}
-static void qdm2_decode (QDM2Context *q, 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);
q->compressed_data = in;
q->compressed_size = q->checksum_size;
-// dump_context(q);
-
/* copy old block, clear new block of output samples */
memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
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,
- uint8_t *buf, int buf_size)
+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,
- .id = CODEC_ID_QDM2,
+ .name = "qdm2",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_QDM2,
.priv_data_size = sizeof(QDM2Context),
- .init = qdm2_decode_init,
- .close = qdm2_decode_close,
- .decode = qdm2_decode_frame,
+ .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