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-rw-r--r--libavcodec/qdm2.c301
1 files changed, 91 insertions, 210 deletions
diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c
index b33e7c64c3..074aafdfb2 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
*/
@@ -44,10 +44,8 @@
#include "mpegaudiodsp.h"
#include "mpegaudio.h"
-#include "qdm2data.h"
#include "qdm2_tablegen.h"
-
#define QDM2_LIST_ADD(list, size, packet) \
do { \
if (size > 0) { \
@@ -165,7 +163,7 @@ 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;
@@ -194,173 +192,11 @@ 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(GetBitContext *gb, VLC *vlc, int flag, int depth)
+static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
{
int value;
@@ -372,7 +208,14 @@ static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
/* 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));
@@ -382,7 +225,7 @@ static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
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);
@@ -691,7 +534,8 @@ 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
+ 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
@@ -802,7 +646,7 @@ static void fill_coding_method_array(sb_int8_array tone_level_idx,
* @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;
@@ -810,14 +654,15 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
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++) {
@@ -841,6 +686,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
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;
}
@@ -865,6 +711,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
}
} 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]];
}
@@ -901,6 +752,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
}
} 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]];
}
@@ -914,6 +770,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
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 {
@@ -926,10 +787,11 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
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);
@@ -945,11 +807,12 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
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);
@@ -986,6 +849,7 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
} // j loop
} // channel loop
} // subband loop
+ return 0;
}
/**
@@ -998,24 +862,27 @@ static void synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
* @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++)
@@ -1024,6 +891,7 @@ static void init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
level += diff;
i += run;
}
+ return 0;
}
/**
@@ -1098,7 +966,7 @@ static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
* @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;
@@ -1116,6 +984,9 @@ static void process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
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));
@@ -1127,6 +998,8 @@ 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;
}
/**
@@ -1196,7 +1069,7 @@ static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
}
-/*
+/**
* Process new subpackets for synthesis filter
*
* @param q context
@@ -1229,7 +1102,7 @@ static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
process_subpacket_12(q, NULL);
}
-/*
+/**
* Decode superblock, fill packet lists.
*
* @param q context
@@ -1389,9 +1262,14 @@ 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(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;
@@ -1704,12 +1582,19 @@ 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;
}
/**
@@ -1722,6 +1607,8 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
int extradata_size;
int tmp_val, tmp, size;
+ qdm2_init_static_data();
+
/* extradata parsing
Structure:
@@ -1810,8 +1697,10 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx)
avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
extradata += 4;
- if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS)
+ 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;
@@ -1838,6 +1727,7 @@ 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;
@@ -1860,18 +1750,9 @@ 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;
@@ -1908,6 +1789,9 @@ 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;
@@ -1979,10 +1863,8 @@ 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++) {
@@ -2003,7 +1885,6 @@ 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 2024-07-09 14:12:15 +0000