[FFmpeg-devel] [PATCH] dcaenc: switch to encode2()
Paul B Mahol
onemda at gmail.com
Thu Mar 22 20:27:19 CET 2012
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
libavcodec/dcaenc.c | 21 ++++++++++++++-------
1 files changed, 14 insertions(+), 7 deletions(-)
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
index 72ea16f..71106d7 100644
--- a/libavcodec/dcaenc.c
+++ b/libavcodec/dcaenc.c
@@ -26,6 +26,7 @@
#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "get_bits.h"
+#include "internal.h"
#include "put_bits.h"
#include "dcaenc.h"
#include "dcadata.h"
@@ -488,14 +489,18 @@ static void put_frame(DCAContext *c,
flush_put_bits(&c->pb);
}
-static int encode_frame(AVCodecContext *avctx, uint8_t *frame,
- int buf_size, void *data)
+static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
int i, k, channel;
DCAContext *c = avctx->priv_data;
- int16_t *samples = data;
- int real_channel = 0;
+ const int16_t *samples;
+ int ret, real_channel = 0;
+ if ((ret = ff_alloc_packet2(avctx, avpkt, DCA_MAX_FRAME_SIZE + DCA_HEADER_SIZE)))
+ return ret;
+
+ samples = (const int16_t *)frame->data[0];
for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
for (channel = 0; channel < c->prim_channels + 1; channel++) {
/* Get 32 PCM samples */
@@ -518,9 +523,11 @@ static int encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
}
- put_frame(c, c->subband, frame);
+ put_frame(c, c->subband, avpkt->data);
- return c->frame_size;
+ avpkt->size = c->frame_size;
+ *got_packet_ptr = 1;
+ return 0;
}
static int encode_init(AVCodecContext *avctx)
@@ -580,7 +587,7 @@ AVCodec ff_dca_encoder = {
.id = CODEC_ID_DTS,
.priv_data_size = sizeof(DCAContext),
.init = encode_init,
- .encode = encode_frame,
+ .encode2 = encode_frame,
.capabilities = CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
--
1.7.7
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