* @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 <stddef.h>
#include <stdio.h>
-#define ALT_BITSTREAM_READER_LE
+#define BITSTREAM_READER_LE
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.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];
* QDM2 decoder context
*/
typedef struct {
+ AVFrame frame;
+
/// 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[1024];
+ float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
/// Synthesis filter
MPADSPContext mpadsp;
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);
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));
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
// dump_context(s);
return 0;
}
}
-static int qdm2_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
QDM2Context *s = avctx->priv_data;
- int16_t *out = data;
- int i;
+ int16_t *out;
+ int i, ret;
if(!buf)
return 0;
if(buf_size < s->checksum_size)
return -1;
- av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
- buf_size, buf, s->checksum_size, data, *data_size);
+ /* get output buffer */
+ s->frame.nb_samples = 16 * s->frame_size;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ out = (int16_t *)s->frame.data[0];
for (i = 0; i < 16; i++) {
if (qdm2_decode(s, buf, out) < 0)
out += s->channels * s->frame_size;
}
- *data_size = (uint8_t*)out - (uint8_t*)data;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
return s->checksum_size;
}
.init = qdm2_decode_init,
.close = qdm2_decode_close,
.decode = qdm2_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
};