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Code Issues Releases Activity

ralf: use planar sample format

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This commit is contained in:
Justin Ruggles 2012-08-28 13:04:49 -04:00
parent 1a3459033d
commit a34be78546
1 changed files with 20 additions and 15 deletions
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35
libavcodec/ralf.c
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@ -149,7 +149,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
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;
@ -338,7 +338,8 @@ static void apply_lpc(RALFContext *ctx, int ch, int length, int bits)
}
}
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;
@ -382,35 +383,35 @@ static int decode_block(AVCodecContext *avctx, GetBitContext *gb, int16_t *dst)
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;
}
@ -424,7 +425,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
RALFContext *ctx = avctx->priv_data;
int16_t *samples;
int16_t *samples0;
int16_t *samples1;
int ret;
GetBitContext gb;
int table_size, table_bytes, i;
@ -465,7 +467,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
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 *)ctx->frame.data[0];
samples1 = (int16_t *)ctx->frame.data[1];
if (src_size < 5) {
av_log(avctx, AV_LOG_ERROR, "too short packets are too short!\n");
@ -498,8 +501,8 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
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;
}
@ -533,4 +536,6 @@ AVCodec ff_ralf_decoder = {
.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 },
};