* 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
*/
/// Parameters built from header parameters, do not change during playback
int group_order; ///< order of frame group
int fft_order; ///< order of FFT (actually fftorder+1)
- int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
int frame_size; ///< size of data frame
int frequency_range;
int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
/// 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;
/* stage-3, optional */
if (flag) {
- int tmp = vlc_stage3_values[value];
+ int tmp;
+
+ if (value >= 60) {
+ av_log(0, 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));
* @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, int length, int sb_min, int sb_max)
+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 joined_stereo, zero_encoding, chs;
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++) {
break;
case 30:
- if (get_bits_left(gb) >= 4)
- samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
- else
+ 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);
run = 1;
type34_predictor = samples[0];
type34_first = 0;
} else {
- samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
+ 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 {
} // 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, GetBitContext *gb)
+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;
}
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) {
+ av_log(0, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
+ return;
+ }
offset = 1;
if (n == 0) {
local_int_4 += local_int_10;
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
{
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
+ float *out = q->output_buffer + channel;
int i;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
/* add samples to output buffer */
- for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
- q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
+ for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
+ out[0] += q->fft.complex[channel][i].re * gain;
+ out[q->channels] += q->fft.complex[channel][i].im * gain;
+ out += 2 * q->channels;
+ }
}
PRINT("checksum_size",q->checksum_size);
PRINT("channels",q->channels);
PRINT("nb_channels",q->nb_channels);
- PRINT("fft_frame_size",q->fft_frame_size);
PRINT("fft_size",q->fft_size);
PRINT("sub_sampling",q->sub_sampling);
PRINT("fft_order",q->fft_order);
avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
extradata += 4;
- if (s->channels > MPA_MAX_CHANNELS)
+ if (s->channels > MPA_MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
return AVERROR_INVALIDDATA;
+ }
avctx->sample_rate = AV_RB32(extradata);
extradata += 4;
}
s->fft_order = av_log2(s->fft_size) + 1;
- s->fft_frame_size = 2 * s->fft_size; // complex has two floats
// 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;
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;
projects / ffmpeg / blobdiff