* Dedicated to the mastermind behind it, Ralph Wiggum.
*/
+#include "libavutil/channel_layout.h"
#include "avcodec.h"
#include "get_bits.h"
#include "golomb.h"
+#include "internal.h"
#include "unary.h"
-#include "libavutil/audioconvert.h"
#include "ralfdata.h"
#define FILTER_NONE 0
#define RALF_MAX_PKT_SIZE 8192
typedef struct RALFContext {
- AVFrame frame;
-
int version;
int max_frame_size;
VLCSet sets[3];
avctx->sample_rate, avctx->channels);
return AVERROR_INVALIDDATA;
}
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO
: AV_CH_LAYOUT_MONO;
- avcodec_get_frame_defaults(&ctx->frame);
- avctx->coded_frame = &ctx->frame;
-
ctx->max_frame_size = AV_RB32(avctx->extradata + 16);
if (ctx->max_frame_size > (1 << 20) || !ctx->max_frame_size) {
av_log(avctx, AV_LOG_ERROR, "invalid frame size %d\n",
}
}
-static int decode_block(AVCodecContext *avctx, GetBitContext *gb, int16_t *dst)
+static int decode_block(AVCodecContext *avctx, GetBitContext *gb,
+ int16_t *dst0, int16_t *dst1)
{
RALFContext *ctx = avctx->priv_data;
int len, ch, ret;
switch (dmode) {
case 0:
for (i = 0; i < len; i++)
- *dst++ = ch0[i] + ctx->bias[0];
+ dst0[i] = ch0[i] + ctx->bias[0];
break;
case 1:
for (i = 0; i < len; i++) {
- *dst++ = ch0[i] + ctx->bias[0];
- *dst++ = ch1[i] + ctx->bias[1];
+ dst0[i] = ch0[i] + ctx->bias[0];
+ dst1[i] = ch1[i] + ctx->bias[1];
}
break;
case 2:
for (i = 0; i < len; i++) {
ch0[i] += ctx->bias[0];
- *dst++ = ch0[i];
- *dst++ = ch0[i] - (ch1[i] + ctx->bias[1]);
+ dst0[i] = ch0[i];
+ dst1[i] = ch0[i] - (ch1[i] + ctx->bias[1]);
}
break;
case 3:
for (i = 0; i < len; i++) {
t = ch0[i] + ctx->bias[0];
t2 = ch1[i] + ctx->bias[1];
- *dst++ = t + t2;
- *dst++ = t;
+ dst0[i] = t + t2;
+ dst1[i] = t;
}
break;
case 4:
for (i = 0; i < len; i++) {
t = ch1[i] + ctx->bias[1];
t2 = ((ch0[i] + ctx->bias[0]) << 1) | (t & 1);
- *dst++ = (t2 + t) / 2;
- *dst++ = (t2 - t) / 2;
+ dst0[i] = (t2 + t) / 2;
+ dst1[i] = (t2 - t) / 2;
}
break;
}
AVPacket *avpkt)
{
RALFContext *ctx = avctx->priv_data;
- int16_t *samples;
+ AVFrame *frame = data;
+ int16_t *samples0;
+ int16_t *samples1;
int ret;
GetBitContext gb;
int table_size, table_bytes, i;
src_size = avpkt->size;
}
- ctx->frame.nb_samples = ctx->max_frame_size;
- if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
+ frame->nb_samples = ctx->max_frame_size;
+ if ((ret = ff_get_buffer(avctx, frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "Me fail get_buffer()? That's unpossible!\n");
return ret;
}
- samples = (int16_t*)ctx->frame.data[0];
+ samples0 = (int16_t *)frame->data[0];
+ samples1 = (int16_t *)frame->data[1];
if (src_size < 5) {
av_log(avctx, AV_LOG_ERROR, "too short packets are too short!\n");
break;
}
init_get_bits(&gb, block_pointer, ctx->block_size[i] * 8);
- if (decode_block(avctx, &gb, samples + ctx->sample_offset
- * avctx->channels) < 0) {
+ if (decode_block(avctx, &gb, samples0 + ctx->sample_offset,
+ samples1 + ctx->sample_offset) < 0) {
av_log(avctx, AV_LOG_ERROR, "Sir, I got carsick in your office. Not decoding the rest of packet.\n");
break;
}
bytes_left -= ctx->block_size[i];
}
- ctx->frame.nb_samples = ctx->sample_offset;
- *got_frame_ptr = ctx->sample_offset > 0;
- *(AVFrame*)data = ctx->frame;
+ frame->nb_samples = ctx->sample_offset;
+ *got_frame_ptr = ctx->sample_offset > 0;
return avpkt->size;
}
.flush = decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("RealAudio Lossless"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE },
};