* 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 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 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,
+ * 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file qdm2.c
+ * @file libavcodec/qdm2.c
* QDM2 decoder
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
- * The decoder is not perfect yet, there are still some distorions expecially
- * on files encoded with 16 or 8 subbands
+ * The decoder is not perfect yet, there are still some distortions
+ * especially on files encoded with 16 or 8 subbands.
*/
#include <math.h>
#define ALT_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 "mpegaudio.h"
#include "qdm2data.h"
} QDM2SubPacket;
/**
- * A node in subpacket list
+ * 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_16(QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
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];
/// Synthesis filter
- MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
+ DECLARE_ALIGNED_16(MPA_INT, 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_16(int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
/// Mixed temporary data used in decoding
float tone_level[MPA_MAX_CHANNELS][30][64];
int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
// Flags
- int has_errors; ///< packet have errors
+ int has_errors; ///< packet has errors
int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
int do_synth_filter; ///< used to perform or skip synthesis filter
int sub_packet;
- int noise_idx; ///< Index for dithering noise table
+ int noise_idx; ///< index for dithering noise table
} QDM2Context;
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() {
+static av_cold void softclip_table_init(void) {
int i;
double dfl = SOFTCLIP_THRESHOLD - 32767;
float delta = 1.0 / -dfl;
// random generated table
-static void rnd_table_init() {
+static av_cold void rnd_table_init(void) {
int i,j;
uint32_t ldw,hdw;
uint64_t tmp64_1;
}
-static void init_noise_samples() {
+static av_cold void init_noise_samples(void) {
int i;
int random_seed = 0;
float delta = 1.0 / 16384.0;
}
}
+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()
+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 const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
* @param length data length
* @param value checksum value
*
- * @return 0 if checksum is ok
+ * @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
+ * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
*
* @param gb bitreader context
* @param sub_packet packet under analysis
sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
}
- av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n",
+ av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
}
/**
- * Return node pointer to first packet of requested type in list
+ * Return node pointer to first packet of requested type in list.
*
- * @param list list of subpacket to be scanned
+ * @param list list of subpackets to be scanned
* @param type type of searched subpacket
* @return node pointer for subpacket if found, else NULL
*/
/**
- * Replaces 8 elements with their average value
- * Called by qdm2_decode_superblock before starting subblocks decoding
+ * Replaces 8 elements with their average value.
+ * Called by qdm2_decode_superblock before starting subblock decoding.
*
* @param q context
*/
/**
- * Build subband samples with noise weighted by q->tone_level
- * Called by synthfilt_build_sb_samples
+ * Build subband samples with noise weighted by q->tone_level.
+ * Called by synthfilt_build_sb_samples.
*
* @param q context
* @param sb subband index
/**
- * Called while processing data from subpackets 11 and 12
- * Used after making changes to coding_method array
+ * Called while processing data from subpackets 11 and 12.
+ * Used after making changes to coding_method array.
*
* @param sb subband index
* @param channels number of channels
* @param coding_method q->coding_method[0][0][0]
*/
- void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
+static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
{
int j,k;
int ch;
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;
* c is built with data from subpacket 11
* Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
*
- * @param tone_level_idx
+ * @param tone_level_idx
* @param tone_level_idx_temp
* @param coding_method q->coding_method[0][0][0]
* @param nb_channels number of channels
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++)
*
* @param q context
* @param gb bitreader context
- * @param length packet length in bit
+ * @param length packet length in bits
* @param sb_min lower subband processed (sb_min included)
* @param sb_max higher subband processed (sb_max excluded)
*/
samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
else
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
-
+
run = 1;
break;
/**
- * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0])
+ * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
* 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
+ * 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 bit
+ * @param length packet length in bits
*/
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
{
if (BITS_LEFT(length,gb) < 16)
break;
diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
-
+
for (k = 1; k <= run; k++)
quantized_coeffs[i + k] = (level + ((k * diff) / run));
-
+
level += diff;
i += run;
}
*
* @param q context
* @param gb bitreader context
- * @param length packet length in bit
+ * @param length packet length in bits
*/
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
{
GetBitContext gb;
int i, j, k, n, ch, run, level, diff;
- init_get_bits(&gb, node->packet->data, node->packet->size);
+ init_get_bits(&gb, node->packet->data, node->packet->size*8);
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
*
* @param q context
* @param node pointer to node with packet
- * @param length packet length in bit
+ * @param length packet length in bits
*/
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
{
GetBitContext gb;
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
+ 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);
{
GetBitContext gb;
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
+ 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 bit
+ * @param length packet length in bits
*/
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
{
GetBitContext gb;
- init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
+ 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));
}
/*
- * Decode superblock, fill packet lists
+ * Decode superblock, fill packet lists.
*
* @param q context
*/
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
- init_get_bits(&gb, q->compressed_data, q->compressed_size);
+ init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
qdm2_decode_sub_packet_header(&gb, &header);
if (header.type < 2 || header.type >= 8) {
q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
- init_get_bits(&gb, header.data, header.size);
+ 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);
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
/* seek to next block */
- init_get_bits(&gb, header.data, header.size);
+ init_get_bits(&gb, header.data, header.size*8);
skip_bits(&gb, next_index*8);
if (next_index >= header.size)
break;
}
- /* decode sub packet */
+ /* decode subpacket */
packet = &q->sub_packets[i];
qdm2_decode_sub_packet_header(&gb, packet);
next_index = packet->size + get_bits_count(&gb) / 8;
packet_bytes -= sub_packet_size;
- /* add sub packet to 'all sub packets' list */
+ /* add subpacket to 'all subpackets' list */
q->sub_packet_list_A[i].packet = packet;
- /* add sub packet to related list */
+ /* add subpacket to related list */
if (packet->type == 8) {
SAMPLES_NEEDED_2("packet type 8");
return;
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 sub packets ordered by type, largest type first */
+ /* 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 sub packet with largest type less than max */
- for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
+ /* find subpacket with largest type less than max */
+ 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;
/* decode FFT tones */
- init_get_bits (&gb, packet->data, packet->size);
+ init_get_bits (&gb, packet->data, packet->size*8);
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
unknown_flag = 1;
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;
-
- /* pre rotation (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;
+ 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]);
/* 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];
+ q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
}
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_window, &dither_state,
samples_ptr, q->nb_channels,
q->sb_samples[ch][(8 * index) + i]);
samples_ptr += 32 * q->nb_channels;
*
* @param q context
*/
-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(ff_mpa_synth_window);
softclip_table_init();
rnd_table_init();
init_noise_samples();
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);
/**
* 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
-
+
Structure:
wave {
frma (QDM2)
QDCA
QDCP
}
-
+
32 size (including this field)
32 tag (=frma)
32 type (=QDM2 or QDMC)
-
+
32 size (including this field, in bytes)
32 tag (=QDCA) // maybe mandatory parameters
32 unknown (=1)
32 block size (=4096)
32 frame size (=256) (for one channel)
32 packet size (=1300)
-
+
32 size (including this field, in bytes)
32 tag (=QDCP) // maybe some tuneable parameters
32 float1 (=1.0)
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;
- 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);
s->fft_order = av_log2(s->fft_size) + 1;
s->fft_frame_size = 2 * s->fft_size; // complex has two floats
s->group_order = av_log2(s->group_size) + 1;
s->frame_size = s->group_size / 16; // 16 iterations per super block
- if (s->fft_order == 8)
- s->sub_sampling = 1;
- else
- s->sub_sampling = 2;
+ s->sub_sampling = s->fft_order - 7;
s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
-
+
switch ((s->sub_sampling * 2 + s->channels - 1)) {
case 0: tmp = 40; break;
case 1: tmp = 48; break;
s->cm_table_select = tmp_val;
if (s->sub_sampling == 0)
- tmp = 16000;
+ tmp = 7999;
else
tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
/*
- 0: 16000 -> 1
+ 0: 7999 -> 0
1: 20000 -> 2
2: 28000 -> 2
*/
else
s->coeff_per_sb_select = 2;
- if (s->fft_order != 8 && s->fft_order != 9)
+ // 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);
-
- 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);
+ return -1;
}
- ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
+ ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
+
qdm2_init(s);
-
+
+ avctx->sample_fmt = SAMPLE_FMT_S16;
+
// dump_context(s);
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;
}
-void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
+static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
{
int ch, i;
const int frame_size = (q->frame_size * q->channels);
-
+
/* select input buffer */
q->compressed_data = in;
q->compressed_size = q->checksum_size;
/* decode block of QDM2 compressed data */
if (q->sub_packet == 0) {
q->has_errors = 0; // zero it for a new super block
- av_log(NULL,AV_LOG_DEBUG,"Super block follows\n");
+ av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
qdm2_decode_super_block(q);
}
- /* parse sub packets */
+ /* parse subpackets */
if (!q->has_errors) {
if (q->sub_packet == 2)
qdm2_decode_fft_packets(q);
static int qdm2_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
- uint8_t *buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
- if((buf == NULL) || (buf_size < s->checksum_size))
+ if(!buf)
return 0;
+ if(buf_size < s->checksum_size)
+ return -1;
*data_size = s->channels * s->frame_size * sizeof(int16_t);
.init = qdm2_decode_init,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
};