* Copyright (c) 2005 Alex Beregszaszi
* Copyright (c) 2005 Roberto Togni
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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 2.1 of the License, or (at your option) any later version.
*
- * Libav 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stddef.h>
#include <stdio.h>
-#define BITSTREAM_READER_LE
#include "libavutil/channel_layout.h"
+
+#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "get_bits.h"
+#include "bytestream.h"
#include "internal.h"
-#include "rdft.h"
-#include "mpegaudiodsp.h"
#include "mpegaudio.h"
+#include "mpegaudiodsp.h"
+#include "rdft.h"
-#include "qdm2data.h"
#include "qdm2_tablegen.h"
-#undef NDEBUG
-#include <assert.h>
-
-
#define QDM2_LIST_ADD(list, size, packet) \
do { \
if (size > 0) { \
/**
* Subpacket
*/
-typedef struct {
+typedef struct QDM2SubPacket {
int type; ///< subpacket type
unsigned int size; ///< subpacket size
const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
} QDM2SubPNode;
-typedef struct {
+typedef struct QDM2Complex {
float re;
float im;
} QDM2Complex;
-typedef struct {
+typedef struct FFTTone {
float level;
QDM2Complex *complex;
const float *table;
short cutoff;
} FFTTone;
-typedef struct {
+typedef struct FFTCoefficient {
int16_t sub_packet;
uint8_t channel;
int16_t offset;
uint8_t phase;
} FFTCoefficient;
-typedef struct {
+typedef struct QDM2FFT {
DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
* QDM2 decoder context
*/
-typedef struct {
+typedef struct QDM2Context {
/// 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[QDM2_MAX_FRAME_SIZE * 2];
+ float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
/// Synthesis filter
MPADSPContext mpadsp;
int noise_idx; ///< index for dithering noise table
} QDM2Context;
-
-static VLC vlc_tab_level;
-static VLC vlc_tab_diff;
-static VLC vlc_tab_run;
-static VLC fft_level_exp_alt_vlc;
-static VLC fft_level_exp_vlc;
-static VLC fft_stereo_exp_vlc;
-static VLC fft_stereo_phase_vlc;
-static VLC vlc_tab_tone_level_idx_hi1;
-static VLC vlc_tab_tone_level_idx_mid;
-static VLC vlc_tab_tone_level_idx_hi2;
-static VLC vlc_tab_type30;
-static VLC vlc_tab_type34;
-static VLC vlc_tab_fft_tone_offset[5];
-
-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 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
};
-static av_cold void qdm2_init_vlc(void)
-{
- static VLC_TYPE qdm2_table[3838][2];
-
- 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);
-}
-
-static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
+static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
{
int value;
/* stage-3, optional */
if (flag) {
- int tmp = vlc_stage3_values[value];
+ int tmp;
+
+ if (value >= 60) {
+ av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
+ return 0;
+ }
+
+ tmp= vlc_stage3_values[value];
if ((value & ~3) > 0)
tmp += get_bits(gb, (value >> 2));
return value;
}
-static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
+static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
{
int value = qdm2_get_vlc(gb, vlc, 0, depth);
/**
* QDM2 checksum
*
- * @param data pointer to data to be checksum'ed
+ * @param data pointer to data to be checksummed
* @param length data length
* @param value checksum value
*
static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
int type)
{
- while (list != NULL && list->packet != NULL) {
+ while (list && list->packet) {
if (list->packet->type == type)
return list;
list = list->next;
if (!superblocktype_2_3) {
/* This case is untested, no samples available */
- SAMPLES_NEEDED
- for (ch = 0; ch < nb_channels; ch++)
+ avpriv_request_sample(NULL, "!superblocktype_2_3");
+ return;
+ for (ch = 0; ch < nb_channels; ch++) {
for (sb = 0; sb < 30; sb++) {
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;
}
tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
}
- acc = 0;
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- acc += tone_level_idx_temp[ch][sb][j];
-
- multres = 0x66666667 * (acc * 10);
- esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++) {
- comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
- if (comp < 0)
- comp += 0xff;
- comp /= 256; // signed shift
- switch(sb) {
- case 0:
- if (comp < 30)
- comp = 30;
- comp += 15;
- break;
- case 1:
- if (comp < 24)
- comp = 24;
- comp += 10;
- break;
- case 2:
- case 3:
- case 4:
- if (comp < 16)
- comp = 16;
- }
- if (comp <= 5)
- tmp = 0;
- else if (comp <= 10)
- tmp = 10;
- else if (comp <= 16)
- tmp = 16;
- else if (comp <= 24)
- tmp = -1;
- else
- tmp = 0;
- coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+ }
+ acc = 0;
+ for (ch = 0; ch < nb_channels; ch++)
+ for (sb = 0; sb < 30; sb++)
+ for (j = 0; j < 64; j++)
+ acc += tone_level_idx_temp[ch][sb][j];
+
+ multres = 0x66666667LL * (acc * 10);
+ esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
+ for (ch = 0; ch < nb_channels; ch++)
+ for (sb = 0; sb < 30; sb++)
+ for (j = 0; j < 64; j++) {
+ comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
+ if (comp < 0)
+ comp += 0xff;
+ comp /= 256; // signed shift
+ switch(sb) {
+ case 0:
+ if (comp < 30)
+ comp = 30;
+ comp += 15;
+ break;
+ case 1:
+ if (comp < 24)
+ comp = 24;
+ comp += 10;
+ break;
+ case 2:
+ case 3:
+ case 4:
+ if (comp < 16)
+ comp = 16;
}
+ if (comp <= 5)
+ tmp = 0;
+ else if (comp <= 10)
+ tmp = 10;
+ else if (comp <= 16)
+ tmp = 16;
+ else if (comp <= 24)
+ tmp = -1;
+ else
+ tmp = 0;
+ coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
+ }
+ for (sb = 0; sb < 30; sb++)
+ fix_coding_method_array(sb, nb_channels, coding_method);
+ for (ch = 0; ch < nb_channels; ch++)
for (sb = 0; sb < 30; sb++)
- fix_coding_method_array(sb, nb_channels, coding_method);
- for (ch = 0; ch < nb_channels; ch++)
- for (sb = 0; sb < 30; sb++)
- for (j = 0; j < 64; j++)
- if (sb >= 10) {
- if (coding_method[ch][sb][j] < 10)
- coding_method[ch][sb][j] = 10;
+ for (j = 0; j < 64; j++)
+ if (sb >= 10) {
+ if (coding_method[ch][sb][j] < 10)
+ coding_method[ch][sb][j] = 10;
+ } else {
+ if (sb >= 2) {
+ if (coding_method[ch][sb][j] < 16)
+ coding_method[ch][sb][j] = 16;
} else {
- if (sb >= 2) {
- if (coding_method[ch][sb][j] < 16)
- coding_method[ch][sb][j] = 16;
- } else {
- if (coding_method[ch][sb][j] < 30)
- coding_method[ch][sb][j] = 30;
- }
+ if (coding_method[ch][sb][j] < 30)
+ coding_method[ch][sb][j] = 30;
}
+ }
} else { // superblocktype_2_3 != 0
for (ch = 0; ch < nb_channels; ch++)
for (sb = 0; sb < 30; sb++)
}
/**
- *
* Called by process_subpacket_11 to process more data from subpacket 11
* with sb 0-8.
* Called by process_subpacket_12 to process data from subpacket 12 with
* @param sb_min lower subband processed (sb_min included)
* @param sb_max higher subband processed (sb_max excluded)
*/
-static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
+static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
int length, int sb_min, int sb_max)
{
int sb, j, k, n, ch, run, channels;
int type34_first;
float type34_div = 0;
float type34_predictor;
- float samples[10], sign_bits[16];
+ float samples[10];
+ int sign_bits[16] = {0};
if (length == 0) {
// If no data use noise
for (sb=sb_min; sb < sb_max; sb++)
build_sb_samples_from_noise(q, sb);
- return;
+ return 0;
}
for (sb = sb_min; sb < sb_max; sb++) {
if (fix_coding_method_array(sb, q->nb_channels,
q->coding_method)) {
+ av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
build_sb_samples_from_noise(q, sb);
continue;
}
}
} else {
n = get_bits(gb, 8);
+ if (n >= 243) {
+ av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
+ return AVERROR_INVALIDDATA;
+ }
+
for (k = 0; k < 5; k++)
samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
}
}
} else {
n = get_bits (gb, 8);
+ if (n >= 243) {
+ av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
+ return AVERROR_INVALIDDATA;
+ }
+
for (k = 0; k < 5; k++)
samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
}
case 24:
if (get_bits_left(gb) >= 7) {
n = get_bits(gb, 7);
+ if (n >= 125) {
+ av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
+ return AVERROR_INVALIDDATA;
+ }
+
for (k = 0; k < 3; k++)
samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
} else {
case 30:
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);
+ if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
+ av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
+ return AVERROR_INVALIDDATA;
+ }
+ samples[0] = type30_dequant[index];
} else
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
type34_first = 0;
} else {
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);
+ if (index >= FF_ARRAY_ELEMS(type34_delta)) {
+ av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
+ return AVERROR_INVALIDDATA;
+ }
+ samples[0] = type34_delta[index] / type34_div + type34_predictor;
+ type34_predictor = samples[0];
}
} else {
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
} // j loop
} // channel loop
} // subband loop
+ return 0;
}
/**
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
* @param gb bitreader context
*/
-static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
+static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
GetBitContext *gb)
{
int i, k, run, level, diff;
if (get_bits_left(gb) < 16)
- return;
+ return -1;
level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
quantized_coeffs[0] = level;
for (i = 0; i < 7; ) {
if (get_bits_left(gb) < 16)
- break;
+ return -1;
run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
+ if (i + run >= 8)
+ return -1;
+
if (get_bits_left(gb) < 16)
- break;
+ return -1;
diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
for (k = 1; k <= run; k++)
level += diff;
i += run;
}
+ return 0;
}
/**
* @param q context
* @param node pointer to node with packet
*/
-static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
+static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
{
GetBitContext gb;
int i, j, k, n, ch, run, level, diff;
run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
+ if (j + run >= 8)
+ return -1;
+
for (k = 1; k <= run; k++)
q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
for (ch = 0; ch < q->nb_channels; ch++)
for (i = 0; i < 8; i++)
q->quantized_coeffs[ch][0][i] = 0;
+
+ return 0;
}
/**
synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
}
-/*
+/**
* Process new subpackets for synthesis filter
*
* @param q context
QDM2SubPNode *nodes[4];
nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
- if (nodes[0] != NULL)
+ if (nodes[0])
process_subpacket_9(q, nodes[0]);
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
- if (nodes[1] != NULL)
+ if (nodes[1])
process_subpacket_10(q, nodes[1]);
else
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)
+ if (nodes[0] && nodes[1] && nodes[2])
process_subpacket_11(q, nodes[2]);
else
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)
+ if (nodes[0] && nodes[1] && nodes[3])
process_subpacket_12(q, nodes[3]);
else
process_subpacket_12(q, NULL);
}
-/*
+/**
* Decode superblock, fill packet lists.
*
* @param q context
}
} // Packet bytes loop
- if (q->sub_packet_list_D[0].packet != NULL) {
+ if (q->sub_packet_list_D[0].packet) {
process_synthesis_subpackets(q, q->sub_packet_list_D);
q->do_synth_filter = 1;
} else if (q->do_synth_filter) {
local_int_10 = 1 << (q->group_order - duration - 1);
offset = 1;
- while (1) {
+ while (get_bits_left(gb)>0) {
if (q->superblocktype_2_3) {
while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
+ if (get_bits_left(gb)<0) {
+ if(local_int_4 < q->group_size)
+ av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
+ return;
+ }
offset = 1;
if (n == 0) {
local_int_4 += local_int_10;
int i, j, min, max, value, type, unknown_flag;
GetBitContext gb;
- if (q->sub_packet_list_B[0].packet == NULL)
+ if (!q->sub_packet_list_B[0].packet)
return;
/* reset minimum indexes for FFT coefficients */
*
* @param q context
*/
-static av_cold void qdm2_init_static_data(AVCodec *codec) {
+static av_cold void qdm2_init_static_data(void) {
+ static int done;
+
+ if(done)
+ return;
+
qdm2_init_vlc();
ff_mpa_synth_init_float(ff_mpa_synth_window_float);
softclip_table_init();
rnd_table_init();
init_noise_samples();
+
+ done = 1;
}
/**
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;
+ GetByteContext gb;
+
+ qdm2_init_static_data();
/* extradata parsing
if (!avctx->extradata || (avctx->extradata_size < 48)) {
av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
- extradata = avctx->extradata;
- extradata_size = avctx->extradata_size;
+ bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
- while (extradata_size > 7) {
- if (!memcmp(extradata, "frmaQDM", 7))
+ while (bytestream2_get_bytes_left(&gb) > 8) {
+ if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
+ (uint64_t)MKBETAG('Q','D','M','2')))
break;
- extradata++;
- extradata_size--;
+ bytestream2_skip(&gb, 1);
}
- if (extradata_size < 12) {
+ if (bytestream2_get_bytes_left(&gb) < 12) {
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
- extradata_size);
- return -1;
- }
-
- if (memcmp(extradata, "frmaQDM", 7)) {
- av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
- return -1;
- }
-
- if (extradata[7] == 'C') {
-// s->is_qdmc = 1;
- av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
- return -1;
+ bytestream2_get_bytes_left(&gb));
+ return AVERROR_INVALIDDATA;
}
- extradata += 8;
- extradata_size -= 8;
-
- size = AV_RB32(extradata);
+ bytestream2_skip(&gb, 8);
+ size = bytestream2_get_be32(&gb);
- if(size > extradata_size){
+ if (size > bytestream2_get_bytes_left(&gb)) {
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
- extradata_size, size);
- return -1;
+ bytestream2_get_bytes_left(&gb), size);
+ return AVERROR_INVALIDDATA;
}
- extradata += 4;
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
- if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
+ if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
- return -1;
+ return AVERROR_INVALIDDATA;
}
- extradata += 8;
+ bytestream2_skip(&gb, 4);
- avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
- extradata += 4;
- if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS)
+ avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
+ if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
return AVERROR_INVALIDDATA;
+ }
avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
AV_CH_LAYOUT_MONO;
- avctx->sample_rate = AV_RB32(extradata);
- extradata += 4;
-
- avctx->bit_rate = AV_RB32(extradata);
- extradata += 4;
-
- s->group_size = AV_RB32(extradata);
- extradata += 4;
-
- s->fft_size = AV_RB32(extradata);
- extradata += 4;
-
- s->checksum_size = AV_RB32(extradata);
+ avctx->sample_rate = bytestream2_get_be32(&gb);
+ avctx->bit_rate = bytestream2_get_be32(&gb);
+ s->group_size = bytestream2_get_be32(&gb);
+ s->fft_size = bytestream2_get_be32(&gb);
+ s->checksum_size = bytestream2_get_be32(&gb);
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;
// 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;
if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
s->cm_table_select = tmp_val;
- if (s->sub_sampling == 0)
- tmp = 7999;
- else
- tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
- /*
- 0: 7999 -> 0
- 1: 20000 -> 2
- 2: 28000 -> 2
- */
- if (tmp < 8000)
+ if (avctx->bit_rate <= 8000)
s->coeff_per_sb_select = 0;
- else if (tmp <= 16000)
+ else if (avctx->bit_rate < 16000)
s->coeff_per_sb_select = 1;
else
s->coeff_per_sb_select = 2;
// 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;
+ avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
+ return AVERROR_PATCHWELCOME;
}
if (s->fft_size != (1 << (s->fft_order - 1))) {
av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
int ch, i;
const int frame_size = (q->frame_size * q->channels);
+ if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
+ return -1;
+
/* select input buffer */
q->compressed_data = in;
q->compressed_size = q->checksum_size;
for (ch = 0; ch < q->channels; ch++) {
qdm2_calculate_fft(q, ch, q->sub_packet);
- if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
+ if (!q->has_errors && q->sub_packet_list_C[0].packet) {
SAMPLES_NEEDED_2("has errors, and C list is not empty")
return -1;
}
q->sub_packet = (q->sub_packet + 1) % 16;
- /* clip and convert output float[] to 16bit signed samples */
+ /* clip and convert output float[] to 16-bit signed samples */
for (i = 0; i < frame_size; i++) {
int value = (int)q->output_buffer[i];
/* get output buffer */
frame->nb_samples = 16 * s->frame_size;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
- }
out = (int16_t *)frame->data[0];
for (i = 0; i < 16; i++) {
- if (qdm2_decode(s, buf, out) < 0)
- return -1;
+ if ((ret = qdm2_decode(s, buf, out)) < 0)
+ return ret;
out += s->channels * s->frame_size;
}
AVCodec ff_qdm2_decoder = {
.name = "qdm2",
+ .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_QDM2,
.priv_data_size = sizeof(QDM2Context),
.init = qdm2_decode_init,
- .init_static_data = qdm2_init_static_data,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
- .capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
+ .capabilities = AV_CODEC_CAP_DR1,
};