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authorMichael Niedermayer <michaelni@gmx.at>2011-12-03 02:08:55 +0100
committerMichael Niedermayer <michaelni@gmx.at>2011-12-03 03:00:30 +0100
commite4de71677f3adeac0f74b89ac8df5d417364df2c (patch)
tree4792dd8d85d24f0f4eaddabb65f6044727907daa /libavcodec/apedec.c
parent12804348f5babf56a315fa01751eea1ffdddf98a (diff)
parentd268b79e3436107c11ee8bcdf9f3645368bb3fcd (diff)
Merge remote-tracking branch 'qatar/master'
* qatar/master: aac_latm: reconfigure decoder on audio specific config changes latmdec: fix audio specific config parsing Add avcodec_decode_audio4(). avcodec: change number of plane pointers from 4 to 8 at next major bump. Update developers documentation with coding conventions. svq1dec: avoid undefined get_bits(0) call ARM: h264dsp_neon cosmetics ARM: make some NEON macros reusable Do not memcpy raw video frames when using null muxer fate: update asf seektest vp8: flush buffers on size changes. doc: improve general documentation for MacOSX asf: use packet dts as approximation of pts asf: do not call av_read_frame rtsp: Initialize the media_type_mask in the rtp guessing demuxer Cleaned up alacenc.c Conflicts: doc/APIchanges doc/developer.texi libavcodec/8svx.c libavcodec/aacdec.c libavcodec/ac3dec.c libavcodec/avcodec.h libavcodec/nellymoserdec.c libavcodec/tta.c libavcodec/utils.c libavcodec/version.h libavcodec/wmadec.c libavformat/asfdec.c tests/ref/seek/lavf_asf Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/apedec.c')
-rw-r--r--libavcodec/apedec.c37
1 files changed, 22 insertions, 15 deletions
diff --git a/libavcodec/apedec.c b/libavcodec/apedec.c
index 10f75212ff..df7529835c 100644
--- a/libavcodec/apedec.c
+++ b/libavcodec/apedec.c
@@ -129,6 +129,7 @@ typedef struct APEPredictor {
/** Decoder context */
typedef struct APEContext {
AVCodecContext *avctx;
+ AVFrame frame;
DSPContext dsp;
int channels;
int samples; ///< samples left to decode in current frame
@@ -215,6 +216,10 @@ static av_cold int ape_decode_init(AVCodecContext *avctx)
dsputil_init(&s->dsp, avctx);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
+
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
filter_alloc_fail:
ape_decode_close(avctx);
@@ -805,16 +810,15 @@ static void ape_unpack_stereo(APEContext *ctx, int count)
}
}
-static int ape_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int ape_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
APEContext *s = avctx->priv_data;
- int16_t *samples = data;
- int i;
- int blockstodecode, out_size;
+ int16_t *samples;
+ int i, ret;
+ int blockstodecode;
int bytes_used = 0;
/* this should never be negative, but bad things will happen if it is, so
@@ -826,7 +830,7 @@ static int ape_decode_frame(AVCodecContext *avctx,
void *tmp_data;
if (!buf_size) {
- *data_size = 0;
+ *got_frame_ptr = 0;
return 0;
}
if (buf_size < 8) {
@@ -874,18 +878,19 @@ static int ape_decode_frame(AVCodecContext *avctx,
}
if (!s->data) {
- *data_size = 0;
+ *got_frame_ptr = 0;
return buf_size;
}
blockstodecode = FFMIN(BLOCKS_PER_LOOP, s->samples);
- out_size = blockstodecode * avctx->channels *
- av_get_bytes_per_sample(avctx->sample_fmt);
- if (*data_size < out_size) {
- av_log(avctx, AV_LOG_ERROR, "Output buffer is too small.\n");
- return AVERROR(EINVAL);
+ /* get output buffer */
+ s->frame.nb_samples = blockstodecode;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
+ samples = (int16_t *)s->frame.data[0];
s->error=0;
@@ -909,7 +914,9 @@ static int ape_decode_frame(AVCodecContext *avctx,
s->samples -= blockstodecode;
- *data_size = out_size;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
return bytes_used;
}
@@ -927,7 +934,7 @@ AVCodec ff_ape_decoder = {
.init = ape_decode_init,
.close = ape_decode_close,
.decode = ape_decode_frame,
- .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY,
+ .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DELAY | CODEC_CAP_DR1,
.flush = ape_flush,
.long_name = NULL_IF_CONFIG_SMALL("Monkey's Audio"),
};