* 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
* 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>
} 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 level;
float *samples_im;
float *samples_re;
- float *table;
+ const float *table;
int phase;
int phase_shift;
int duration;
} QDM2Complex;
typedef struct {
- QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
+ DECLARE_ALIGNED_16(QDM2Complex, complex[256 + 1]);
float samples_im[MPA_MAX_CHANNELS][256];
float samples_re[MPA_MAX_CHANNELS][256];
} QDM2FFT;
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 DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
-static void softclip_table_init() {
+static 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 void rnd_table_init(void) {
int i,j;
uint32_t ldw,hdw;
uint64_t tmp64_1;
}
-static void init_noise_samples() {
+static void init_noise_samples(void) {
int i;
int random_seed = 0;
float delta = 1.0 / 16384.0;
}
-static void qdm2_init_vlc()
+static void qdm2_init_vlc(void)
{
init_vlc (&vlc_tab_level, 8, 24,
vlc_tab_level_huffbits, 1, 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;
*
* @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)
{
*
* @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)
{
*
* @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)
{
/*
- * Decode superblock, fill packet lists
+ * Decode superblock, fill packet lists.
*
* @param q context
*/
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;
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)
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.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;
float c, s, f0, f1, f2, f3;
int i, j;
- /* pre rotation (or something like that) */
+ /* prerotation (or something like that) */
for (i=1; i < n2; i++) {
j = (n - i);
c = q->exptab[i].re;
*
* @param q context
*/
-void qdm2_init(QDM2Context *q) {
- static int inited = 0;
+static 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);
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);
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);
+ s->checksum_size = AV_RB32(extradata);
extradata += 4;
s->fft_order = av_log2(s->fft_size) + 1;
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 it's > 9 it can overflow s->exptab[]
+ 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);
s->exptab[i].im = sin(alpha);
}
- ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
qdm2_init(s);
-
+
+ avctx->sample_fmt = SAMPLE_FMT_S16;
+
// dump_context(s);
return 0;
}
QDM2Context *s = avctx->priv_data;
ff_fft_end(&s->fft_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)
+ const uint8_t *buf, int buf_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"),
};
projects / ffmpeg / blobdiff