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-rw-r--r--libavcodec/qdm2.c566
1 files changed, 211 insertions, 355 deletions
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index 668e513a66..88b6b19d11 100644
--- a/libavcodec/qdm2.c
+++ b/libavcodec/qdm2.c
@@ -5,20 +5,20 @@
* 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
*/
@@ -39,17 +39,15 @@
#define BITSTREAM_READER_LE
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
+#include "bytestream.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#include "rdft.h"
-#include "vlc.h"
-#include "qdm2data.h"
#include "qdm2_tablegen.h"
-
#define QDM2_LIST_ADD(list, size, packet) \
do { \
if (size > 0) { \
@@ -167,7 +165,7 @@ typedef struct QDM2Context {
/// 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;
@@ -196,197 +194,42 @@ typedef struct QDM2Context {
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(BitstreamContext *bc, VLC *vlc, int flag, int depth)
+static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
{
int value;
- value = bitstream_read_vlc(bc, vlc->table, vlc->bits, depth);
+ value = get_vlc2(gb, vlc->table, vlc->bits, depth);
/* stage-2, 3 bits exponent escape sequence */
if (value-- == 0)
- value = bitstream_read(bc, bitstream_read(bc, 3) + 1);
+ value = get_bits(gb, get_bits(gb, 3) + 1);
/* 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 += bitstream_read(bc, value >> 2);
+ tmp += get_bits(gb, (value >> 2));
value = tmp;
}
return value;
}
-static int qdm2_get_se_vlc(VLC *vlc, BitstreamContext *bc, int depth)
+static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
{
- int value = qdm2_get_vlc(bc, vlc, 0, depth);
+ int value = qdm2_get_vlc(gb, vlc, 0, depth);
return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
}
@@ -413,35 +256,35 @@ static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
/**
* Fill a QDM2SubPacket structure with packet type, size, and data pointer.
*
- * @param bc bitreader context
+ * @param gb bitreader context
* @param sub_packet packet under analysis
*/
-static void qdm2_decode_sub_packet_header(BitstreamContext *bc,
+static void qdm2_decode_sub_packet_header(GetBitContext *gb,
QDM2SubPacket *sub_packet)
{
- sub_packet->type = bitstream_read(bc, 8);
+ sub_packet->type = get_bits(gb, 8);
if (sub_packet->type == 0) {
sub_packet->size = 0;
sub_packet->data = NULL;
} else {
- sub_packet->size = bitstream_read(bc, 8);
+ sub_packet->size = get_bits(gb, 8);
if (sub_packet->type & 0x80) {
sub_packet->size <<= 8;
- sub_packet->size |= bitstream_read(bc, 8);
+ sub_packet->size |= get_bits(gb, 8);
sub_packet->type &= 0x7f;
}
if (sub_packet->type == 0x7f)
- sub_packet->type |= bitstream_read(bc, 8) << 8;
+ sub_packet->type |= (get_bits(gb, 8) << 8);
// FIXME: this depends on bitreader-internal data
- sub_packet->data = &bc->buffer[bitstream_tell(bc) / 8];
+ sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
}
av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
- sub_packet->type, sub_packet->size, bitstream_tell(bc) / 8);
+ sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
}
/**
@@ -693,8 +536,9 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
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;
@@ -723,7 +567,7 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
}
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++)
@@ -800,12 +644,12 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
* sb 8-sb_used.
*
* @param q context
- * @param bc bitreader context
+ * @param gb bitreader context
* @param length packet length in bits
* @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, BitstreamContext *bc,
+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;
@@ -813,14 +657,15 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
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++) {
@@ -831,12 +676,12 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
else if (sb >= 24)
joined_stereo = 1;
else
- joined_stereo = (bitstream_bits_left(bc) >= 1) ? bitstream_read_bit(bc) : 0;
+ joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
if (joined_stereo) {
- if (bitstream_bits_left(bc) >= 16)
+ if (get_bits_left(gb) >= 16)
for (j = 0; j < 16; j++)
- sign_bits[j] = bitstream_read_bit(bc);
+ sign_bits[j] = get_bits1(gb);
for (j = 0; j < 64; j++)
if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
@@ -844,6 +689,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
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;
}
@@ -852,22 +698,27 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
for (ch = 0; ch < channels; ch++) {
FIX_NOISE_IDX(q->noise_idx);
- zero_encoding = (bitstream_bits_left(bc) >= 1) ? bitstream_read_bit(bc) : 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 (bitstream_bits_left(bc) >= 10) {
+ if (get_bits_left(gb) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + 2 * k) >= 128)
break;
- samples[2 * k] = bitstream_read_bit(bc) ? dequant_1bit[joined_stereo][2 * bitstream_read_bit(bc)] : 0;
+ samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
}
} else {
- n = bitstream_read(bc, 8);
+ 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]];
}
@@ -881,10 +732,10 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
break;
case 10:
- if (bitstream_bits_left(bc) >= 1) {
+ if (get_bits_left(gb) >= 1) {
float f = 0.81;
- if (bitstream_read_bit(bc))
+ if (get_bits1(gb))
f = -f;
f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
samples[0] = f;
@@ -895,15 +746,20 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
break;
case 16:
- if (bitstream_bits_left(bc) >= 10) {
+ if (get_bits_left(gb) >= 10) {
if (zero_encoding) {
for (k = 0; k < 5; k++) {
if ((j + k) >= 128)
break;
- samples[k] = (bitstream_read_bit(bc) == 0) ? 0 : dequant_1bit[joined_stereo][2 * bitstream_read_bit(bc)];
+ samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
}
} else {
- n = bitstream_read (bc, 8);
+ 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]];
}
@@ -915,8 +771,13 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
break;
case 24:
- if (bitstream_bits_left(bc) >= 7) {
- n = bitstream_read(bc, 7);
+ 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 {
@@ -927,12 +788,13 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
break;
case 30:
- if (bitstream_bits_left(bc) >= 4) {
- unsigned index = qdm2_get_vlc(bc, &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 (get_bits_left(gb) >= 4) {
+ unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
+ 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);
@@ -940,19 +802,20 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
break;
case 34:
- if (bitstream_bits_left(bc) >= 7) {
+ if (get_bits_left(gb) >= 7) {
if (type34_first) {
- type34_div = (float)(1 << bitstream_read(bc, 2));
- samples[0] = ((float)bitstream_read(bc, 5) - 16.0) / 15.0;
+ 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 {
- unsigned index = qdm2_get_vlc(bc, &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);
+ unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
+ 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);
@@ -989,6 +852,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
} // j loop
} // channel loop
} // subband loop
+ return 0;
}
/**
@@ -999,27 +863,30 @@ static void synthfilt_build_sb_samples(QDM2Context *q, BitstreamContext *bc,
* same VLC tables as process_subpacket_9 are used.
*
* @param quantized_coeffs pointer to quantized_coeffs[ch][0]
- * @param bc bitreader context
+ * @param gb bitreader context
*/
-static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
- BitstreamContext *bc)
+static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
+ GetBitContext *gb)
{
int i, k, run, level, diff;
- if (bitstream_bits_left(bc) < 16)
- return;
- level = qdm2_get_vlc(bc, &vlc_tab_level, 0, 2);
+ if (get_bits_left(gb) < 16)
+ return -1;
+ level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
quantized_coeffs[0] = level;
for (i = 0; i < 7; ) {
- if (bitstream_bits_left(bc) < 16)
- break;
- run = qdm2_get_vlc(bc, &vlc_tab_run, 0, 1) + 1;
+ if (get_bits_left(gb) < 16)
+ return -1;
+ run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
- if (bitstream_bits_left(bc) < 16)
- break;
- diff = qdm2_get_se_vlc(&vlc_tab_diff, bc, 2);
+ if (i + run >= 8)
+ return -1;
+
+ if (get_bits_left(gb) < 16)
+ return -1;
+ diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
for (k = 1; k <= run; k++)
quantized_coeffs[i + k] = (level + ((k * diff) / run));
@@ -1027,6 +894,7 @@ static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
level += diff;
i += run;
}
+ return 0;
}
/**
@@ -1036,16 +904,16 @@ static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
* data from packet 10
*
* @param q context
- * @param bc bitreader context
+ * @param gb bitreader context
*/
-static void init_tone_level_dequantization(QDM2Context *q, BitstreamContext *bc)
+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], bc);
+ init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
- if (bitstream_bits_left(bc) < 16) {
+ if (get_bits_left(gb) < 16) {
memset(q->quantized_coeffs[ch][0], 0, 8);
break;
}
@@ -1056,13 +924,13 @@ static void init_tone_level_dequantization(QDM2Context *q, BitstreamContext *bc)
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (bitstream_bits_left(bc) < 1)
+ if (get_bits_left(gb) < 1)
break;
- if (bitstream_read_bit(bc)) {
+ if (get_bits1(gb)) {
for (k=0; k < 8; k++) {
- if (bitstream_bits_left(bc) < 16)
+ if (get_bits_left(gb) < 16)
break;
- q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_hi1, 0, 2);
+ q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
}
} else {
for (k=0; k < 8; k++)
@@ -1074,9 +942,9 @@ static void init_tone_level_dequantization(QDM2Context *q, BitstreamContext *bc)
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++) {
- if (bitstream_bits_left(bc) < 16)
+ if (get_bits_left(gb) < 16)
break;
- q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_hi2, 0, 2);
+ q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
if (sb > 19)
q->tone_level_idx_hi2[ch][sb] -= 16;
else
@@ -1089,9 +957,9 @@ static void init_tone_level_dequantization(QDM2Context *q, BitstreamContext *bc)
for (sb = 0; sb < n; sb++)
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 8; j++) {
- if (bitstream_bits_left(bc) < 16)
+ if (get_bits_left(gb) < 16)
break;
- q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(bc, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
+ q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
}
}
@@ -1101,23 +969,26 @@ static void init_tone_level_dequantization(QDM2Context *q, BitstreamContext *bc)
* @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)
{
- BitstreamContext bc;
+ GetBitContext gb;
int i, j, k, n, ch, run, level, diff;
- bitstream_init8(&bc, 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;
for (i = 1; i < n; i++)
for (ch = 0; ch < q->nb_channels; ch++) {
- level = qdm2_get_vlc(&bc, &vlc_tab_level, 0, 2);
+ level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
q->quantized_coeffs[ch][i][0] = level;
for (j = 0; j < (8 - 1); ) {
- run = qdm2_get_vlc(&bc, &vlc_tab_run, 0, 1) + 1;
- diff = qdm2_get_se_vlc(&vlc_tab_diff, &bc, 2);
+ 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));
@@ -1130,6 +1001,8 @@ static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
for (ch = 0; ch < q->nb_channels; ch++)
for (i = 0; i < 8; i++)
q->quantized_coeffs[ch][0][i] = 0;
+
+ return 0;
}
/**
@@ -1140,11 +1013,11 @@ static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
*/
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
{
- BitstreamContext bc;
+ GetBitContext gb;
if (node) {
- bitstream_init8(&bc, node->packet->data, node->packet->size);
- init_tone_level_dequantization(q, &bc);
+ 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);
@@ -1159,16 +1032,16 @@ static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
*/
static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
{
- BitstreamContext bc;
+ GetBitContext gb;
int length = 0;
if (node) {
length = node->packet->size * 8;
- bitstream_init(&bc, node->packet->data, length);
+ init_get_bits(&gb, node->packet->data, length);
}
if (length >= 32) {
- int c = bitstream_read(&bc, 13);
+ int c = get_bits(&gb, 13);
if (c > 3)
fill_coding_method_array(q->tone_level_idx,
@@ -1177,7 +1050,7 @@ static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
q->superblocktype_2_3, q->cm_table_select);
}
- synthfilt_build_sb_samples(q, &bc, length, 0, 8);
+ synthfilt_build_sb_samples(q, &gb, length, 0, 8);
}
/**
@@ -1188,18 +1061,18 @@ static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
*/
static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
{
- BitstreamContext bc;
+ GetBitContext gb;
int length = 0;
if (node) {
length = node->packet->size * 8;
- bitstream_init(&bc, node->packet->data, length);
+ init_get_bits(&gb, node->packet->data, length);
}
- synthfilt_build_sb_samples(q, &bc, length, 8, QDM2_SB_USED(q->sub_sampling));
+ synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
}
-/*
+/**
* Process new subpackets for synthesis filter
*
* @param q context
@@ -1232,14 +1105,14 @@ static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
process_subpacket_12(q, NULL);
}
-/*
+/**
* Decode superblock, fill packet lists.
*
* @param q context
*/
static void qdm2_decode_super_block(QDM2Context *q)
{
- BitstreamContext bc;
+ GetBitContext gb;
QDM2SubPacket header, *packet;
int i, packet_bytes, sub_packet_size, sub_packets_D;
unsigned int next_index = 0;
@@ -1253,8 +1126,8 @@ static void qdm2_decode_super_block(QDM2Context *q)
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
- bitstream_init8(&bc, q->compressed_data, q->compressed_size);
- qdm2_decode_sub_packet_header(&bc, &header);
+ 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->has_errors = 1;
@@ -1263,13 +1136,13 @@ static void qdm2_decode_super_block(QDM2Context *q)
}
q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
- packet_bytes = (q->compressed_size - bitstream_tell(&bc) / 8);
+ packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
- bitstream_init8(&bc, 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 * bitstream_read(&bc, 8);
- csum += 2 * bitstream_read(&bc, 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);
@@ -1301,8 +1174,8 @@ static void qdm2_decode_super_block(QDM2Context *q)
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
/* seek to next block */
- bitstream_init8(&bc, header.data, header.size);
- bitstream_skip(&bc, next_index * 8);
+ init_get_bits(&gb, header.data, header.size * 8);
+ skip_bits(&gb, next_index * 8);
if (next_index >= header.size)
break;
@@ -1310,8 +1183,8 @@ static void qdm2_decode_super_block(QDM2Context *q)
/* decode subpacket */
packet = &q->sub_packets[i];
- qdm2_decode_sub_packet_header(&bc, packet);
- next_index = packet->size + bitstream_tell(&bc) / 8;
+ qdm2_decode_sub_packet_header(&gb, packet);
+ next_index = packet->size + get_bits_count(&gb) / 8;
sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
if (packet->type == 0)
@@ -1337,10 +1210,10 @@ static void qdm2_decode_super_block(QDM2Context *q)
QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
} else if (packet->type == 13) {
for (j = 0; j < 6; j++)
- q->fft_level_exp[j] = bitstream_read(&bc, 6);
+ q->fft_level_exp[j] = get_bits(&gb, 6);
} else if (packet->type == 14) {
for (j = 0; j < 6; j++)
- q->fft_level_exp[j] = qdm2_get_vlc(&bc, &fft_level_exp_vlc, 0, 2);
+ q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
} else if (packet->type == 15) {
SAMPLES_NEEDED_2("packet type 15")
return;
@@ -1378,7 +1251,7 @@ static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
}
static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
- BitstreamContext *bc, int b)
+ GetBitContext *gb, int b)
{
int channel, stereo, phase, exp;
int local_int_4, local_int_8, stereo_phase, local_int_10;
@@ -1392,9 +1265,14 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
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(bc, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
+ 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;
@@ -1406,7 +1284,7 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
}
offset += (n - 2);
} else {
- offset += qdm2_get_vlc(bc, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
+ offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
while (offset >= (local_int_10 - 1)) {
offset += (1 - (local_int_10 - 1));
local_int_4 += local_int_10;
@@ -1422,24 +1300,24 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
return;
if (q->nb_channels > 1) {
- channel = bitstream_read_bit(bc);
- stereo = bitstream_read_bit(bc);
+ channel = get_bits1(gb);
+ stereo = get_bits1(gb);
} else {
channel = 0;
stereo = 0;
}
- exp = qdm2_get_vlc(bc, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
+ exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
exp = (exp < 0) ? 0 : exp;
- phase = bitstream_read(bc, 3);
+ phase = get_bits(gb, 3);
stereo_exp = 0;
stereo_phase = 0;
if (stereo) {
- stereo_exp = (exp - qdm2_get_vlc(bc, &fft_stereo_exp_vlc, 0, 1));
- stereo_phase = (phase - qdm2_get_vlc(bc, &fft_stereo_phase_vlc, 0, 1));
+ stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
+ stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
if (stereo_phase < 0)
stereo_phase += 8;
}
@@ -1461,7 +1339,7 @@ static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
static void qdm2_decode_fft_packets(QDM2Context *q)
{
int i, j, min, max, value, type, unknown_flag;
- BitstreamContext bc;
+ GetBitContext gb;
if (!q->sub_packet_list_B[0].packet)
return;
@@ -1496,7 +1374,7 @@ static void qdm2_decode_fft_packets(QDM2Context *q)
return;
/* decode FFT tones */
- bitstream_init8(&bc, 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;
@@ -1509,15 +1387,15 @@ static void qdm2_decode_fft_packets(QDM2Context *q)
int duration = q->sub_sampling + 5 - (type & 15);
if (duration >= 0 && duration < 4)
- qdm2_fft_decode_tones(q, duration, &bc, unknown_flag);
+ qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
} else if (type == 31) {
for (j = 0; j < 4; j++)
- qdm2_fft_decode_tones(q, j, &bc, unknown_flag);
+ qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
} else if (type == 46) {
for (j = 0; j < 6; j++)
- q->fft_level_exp[j] = bitstream_read(&bc, 6);
+ q->fft_level_exp[j] = get_bits(&gb, 6);
for (j = 0; j < 4; j++)
- qdm2_fft_decode_tones(q, j, &bc, unknown_flag);
+ qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
}
} // Loop on B packets
@@ -1707,12 +1585,19 @@ static void qdm2_synthesis_filter(QDM2Context *q, int index)
*
* @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;
}
/**
@@ -1721,9 +1606,10 @@ static av_cold void qdm2_init_static_data(AVCodec *codec) {
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
@@ -1764,73 +1650,51 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
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);
+ bytestream2_get_bytes_left(&gb));
return AVERROR_INVALIDDATA;
}
- if (memcmp(extradata, "frmaQDM", 7)) {
- av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
- return AVERROR_INVALIDDATA;
- }
+ bytestream2_skip(&gb, 8);
+ size = bytestream2_get_be32(&gb);
- if (extradata[7] == 'C') {
-// s->is_qdmc = 1;
- avpriv_report_missing_feature(avctx, "QDMC version 1");
- return AVERROR_PATCHWELCOME;
- }
-
- extradata += 8;
- extradata_size -= 8;
-
- size = AV_RB32(extradata);
-
- 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);
+ 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 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;
@@ -1841,6 +1705,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
// 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;
@@ -1863,18 +1728,9 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
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;
@@ -1911,6 +1767,9 @@ static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
int ch, i;
const int frame_size = (q->frame_size * q->channels);
+ if((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;
@@ -1982,10 +1841,8 @@ static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
/* 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++) {
@@ -2006,7 +1863,6 @@ AVCodec ff_qdm2_decoder = {
.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 = AV_CODEC_CAP_DR1,
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-14 18:23:28 +0000