* available.
*/
+#include "libavutil/channel_layout.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "bytestream.h"
#include "fft.h"
-#include "libavutil/audioconvert.h"
+#include "internal.h"
#include "sinewin.h"
#include "cookdata.h"
int size;
int num_channels;
int cookversion;
- int samples_per_frame;
int subbands;
int js_subband_start;
int js_vlc_bits;
AVCodecContext* avctx;
DSPContext dsp;
- AVFrame frame;
GetBitContext gb;
/* stream data */
int num_vectors;
static int cook_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
+ AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data;
/* get output buffer */
if (q->discarded_packets >= 2) {
- q->frame.nb_samples = q->samples_per_channel;
- if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
+ frame->nb_samples = q->samples_per_channel;
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- samples = (float **)q->frame.extended_data;
+ samples = (float **)frame->extended_data;
}
/* estimate subpacket sizes */
return avctx->block_align;
}
- *got_frame_ptr = 1;
- *(AVFrame *) data = q->frame;
+ *got_frame_ptr = 1;
return avctx->block_align;
}
static void dump_cook_context(COOKContext *q)
{
//int i=0;
-#define PRINT(a, b) av_log(q->avctx, AV_LOG_ERROR, " %s = %d\n", a, b);
- av_log(q->avctx, AV_LOG_ERROR, "COOKextradata\n");
- av_log(q->avctx, AV_LOG_ERROR, "cookversion=%x\n", q->subpacket[0].cookversion);
+#define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
+ av_dlog(q->avctx, "COOKextradata\n");
+ av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
if (q->subpacket[0].cookversion > STEREO) {
PRINT("js_subband_start", q->subpacket[0].js_subband_start);
PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
}
- av_log(q->avctx, AV_LOG_ERROR, "COOKContext\n");
+ av_dlog(q->avctx, "COOKContext\n");
PRINT("nb_channels", q->avctx->channels);
PRINT("bit_rate", q->avctx->bit_rate);
PRINT("sample_rate", q->avctx->sample_rate);
PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
- PRINT("samples_per_frame", q->subpacket[0].samples_per_frame);
PRINT("subbands", q->subpacket[0].subbands);
PRINT("js_subband_start", q->subpacket[0].js_subband_start);
PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
int extradata_size = avctx->extradata_size;
int s = 0;
unsigned int channel_mask = 0;
+ int samples_per_frame;
int ret;
q->avctx = avctx;
Swap to right endianness so we don't need to care later on. */
if (extradata_size >= 8) {
q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
- q->subpacket[s].samples_per_frame = bytestream_get_be16(&edata_ptr);
+ samples_per_frame = bytestream_get_be16(&edata_ptr);
q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
extradata_size -= 8;
}
}
/* Initialize extradata related variables. */
- q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame / avctx->channels;
+ q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
/* Initialize default data states. */
switch (q->subpacket[s].cookversion) {
case MONO:
if (avctx->channels != 1) {
- av_log_ask_for_sample(avctx, "Container channels != 1.\n");
+ avpriv_request_sample(avctx, "Container channels != 1");
return AVERROR_PATCHWELCOME;
}
av_log(avctx, AV_LOG_DEBUG, "MONO\n");
break;
case JOINT_STEREO:
if (avctx->channels != 2) {
- av_log_ask_for_sample(avctx, "Container channels != 2.\n");
+ avpriv_request_sample(avctx, "Container channels != 2");
return AVERROR_PATCHWELCOME;
}
av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
q->subpacket[s].js_subband_start;
q->subpacket[s].joint_stereo = 1;
q->subpacket[s].num_channels = 2;
- q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame >> 1;
+ q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
if (q->subpacket[s].samples_per_channel > 256) {
q->subpacket[s].log2_numvector_size = 6;
q->subpacket[s].log2_numvector_size = 7;
}
} else
- q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame;
+ q->subpacket[s].samples_per_channel = samples_per_frame;
break;
default:
- av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
+ avpriv_request_sample(avctx, "Cook version %d",
+ q->subpacket[s].cookversion);
return AVERROR_PATCHWELCOME;
}
/* Try to catch some obviously faulty streams, othervise it might be exploitable */
if (q->subpacket[s].total_subbands > 53) {
- av_log_ask_for_sample(avctx, "total_subbands > 53\n");
+ avpriv_request_sample(avctx, "total_subbands > 53");
return AVERROR_PATCHWELCOME;
}
}
if (q->subpacket[s].subbands > 50) {
- av_log_ask_for_sample(avctx, "subbands > 50\n");
+ avpriv_request_sample(avctx, "subbands > 50");
return AVERROR_PATCHWELCOME;
}
q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
q->num_subpackets++;
s++;
if (s > MAX_SUBPACKETS) {
- av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
+ avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS);
return AVERROR_PATCHWELCOME;
}
}
/* Try to catch some obviously faulty streams, othervise it might be exploitable */
if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
q->samples_per_channel != 1024) {
- av_log_ask_for_sample(avctx,
- "unknown amount of samples_per_channel = %d\n",
+ avpriv_request_sample(avctx, "samples_per_channel = %d",
q->samples_per_channel);
return AVERROR_PATCHWELCOME;
}
else
avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
- avcodec_get_frame_defaults(&q->frame);
- avctx->coded_frame = &q->frame;
-
#ifdef DEBUG
dump_cook_context(q);
#endif
AVCodec ff_cook_decoder = {
.name = "cook",
+ .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_COOK,
.priv_data_size = sizeof(COOKContext),
.close = cook_decode_close,
.decode = cook_decode_frame,
.capabilities = CODEC_CAP_DR1,
- .long_name = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};