* Copyright (c) 2005 Alex Beregszaszi
* Copyright (c) 2005 Roberto Togni
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* @file
* QDM2 decoder
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
+ *
* The decoder is not perfect yet, there are still some distortions
* especially on files encoded with 16 or 8 subbands.
*/
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
-#include "fft.h"
+#include "rdft.h"
+#include "mpegaudiodsp.h"
#include "mpegaudio.h"
#include "qdm2data.h"
#define SAMPLES_NEEDED_2(why) \
av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
+#define QDM2_MAX_FRAME_SIZE 512
typedef int8_t sb_int8_array[2][30][64];
} FFTCoefficient;
typedef struct {
- DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
+ DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
} QDM2FFT;
/**
/// I/O data
const uint8_t *compressed_data;
int compressed_size;
- float output_buffer[1024];
+ float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
/// Synthesis filter
- DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
+ MPADSPContext mpadsp;
+ DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
int synth_buf_offset[MPA_MAX_CHANNELS];
- DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+ DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
+ DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
/// Mixed temporary data used in decoding
float tone_level[MPA_MAX_CHANNELS][30][64];
}
}
-
-/* for floating point to fixed point conversion */
-static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
-
-
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
{
int value;
for (ch = 0; ch < q->nb_channels; ch++)
for (j = 0; j < 64; j++) {
- q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
- q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
+ q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
+ q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
}
}
for (chs = 0; chs < q->nb_channels; chs++)
for (k = 0; k < run; k++)
if ((j + k) < 128)
- q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
+ q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
} else {
for (k = 0; k < run; k++)
if ((j + k) < 128)
- q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
+ q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
}
j += run;
return;
local_int_14 = (offset >> local_int_8);
+ if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
+ return;
if (q->nb_channels > 1) {
channel = get_bits1(gb);
int i;
q->fft.complex[channel][0].re *= 2.0f;
q->fft.complex[channel][0].im = 0.0f;
- ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
+ 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;
*/
static void qdm2_synthesis_filter (QDM2Context *q, int index)
{
- OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
int i, k, ch, sb_used, sub_sampling, dither_state = 0;
/* copy sb_samples */
q->sb_samples[ch][(8 * index) + i][k] = 0;
for (ch = 0; ch < q->nb_channels; ch++) {
- OUT_INT *samples_ptr = samples + ch;
+ float *samples_ptr = q->samples + ch;
for (i = 0; i < 8; i++) {
- ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
- ff_mpa_synth_window, &dither_state,
+ ff_mpa_synth_filter_float(&q->mpadsp,
+ q->synth_buf[ch], &(q->synth_buf_offset[ch]),
+ ff_mpa_synth_window_float, &dither_state,
samples_ptr, q->nb_channels,
q->sb_samples[ch][(8 * index) + i]);
samples_ptr += 32 * q->nb_channels;
for (ch = 0; ch < q->channels; ch++)
for (i = 0; i < q->frame_size; i++)
- q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
+ q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
}
initialized = 1;
qdm2_init_vlc();
- ff_mpa_synth_init(ff_mpa_synth_window);
+ ff_mpa_synth_init_float(ff_mpa_synth_window_float);
softclip_table_init();
rnd_table_init();
init_noise_samples();
avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
extradata += 4;
+ if (s->channels > MPA_MAX_CHANNELS)
+ return AVERROR_INVALIDDATA;
avctx->sample_rate = AV_RB32(extradata);
extradata += 4;
// 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;
s->sub_sampling = s->fft_order - 7;
s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
}
ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
+ ff_mpadsp_init(&s->mpadsp);
qdm2_init(s);
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
int16_t *out = data;
- int i;
+ int i, out_size;
if(!buf)
return 0;
if(buf_size < s->checksum_size)
return -1;
+ out_size = 16 * s->channels * s->frame_size *
+ av_get_bytes_per_sample(avctx->sample_fmt);
+ if (*data_size < out_size) {
+ av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
+ return AVERROR(EINVAL);
+ }
+
av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
buf_size, buf, s->checksum_size, data, *data_size);
out += s->channels * s->frame_size;
}
- *data_size = (uint8_t*)out - (uint8_t*)data;
+ *data_size = out_size;
- return buf_size;
+ return s->checksum_size;
}
-AVCodec qdm2_decoder =
+AVCodec ff_qdm2_decoder =
{
.name = "qdm2",
.type = AVMEDIA_TYPE_AUDIO,
projects / ffmpeg / blobdiff