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authorJustin Ruggles <justin.ruggles@gmail.com>2011-09-06 12:17:45 -0400
committerJustin Ruggles <justin.ruggles@gmail.com>2011-12-02 17:40:40 -0500
commit0eea212943544d40f99b05571aa7159d78667154 (patch)
tree1e6b0271a633bf8a3f92c78bdfbaca275498ee26 /libavcodec/flacdec.c
parent560f773c7ddd17f66e2621222980c1359a9027be (diff)
Add avcodec_decode_audio4().
Deprecate avcodec_decode_audio3(). Implement audio support in avcodec_default_get_buffer(). Implement the new audio decoder API in all audio decoders.
Diffstat (limited to 'libavcodec/flacdec.c')
-rw-r--r--libavcodec/flacdec.c37
1 files changed, 21 insertions, 16 deletions
diff --git a/libavcodec/flacdec.c b/libavcodec/flacdec.c
index 95cf2bccb4..58eb66def9 100644
--- a/libavcodec/flacdec.c
+++ b/libavcodec/flacdec.c
@@ -49,6 +49,7 @@ typedef struct FLACContext {
FLACSTREAMINFO
AVCodecContext *avctx; ///< parent AVCodecContext
+ AVFrame frame;
GetBitContext gb; ///< GetBitContext initialized to start at the current frame
int blocksize; ///< number of samples in the current frame
@@ -116,6 +117,9 @@ static av_cold int flac_decode_init(AVCodecContext *avctx)
allocate_buffers(s);
s->got_streaminfo = 1;
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
return 0;
}
@@ -542,20 +546,18 @@ static int decode_frame(FLACContext *s)
return 0;
}
-static int flac_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+static int flac_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
FLACContext *s = avctx->priv_data;
int i, j = 0, bytes_read = 0;
- int16_t *samples_16 = data;
- int32_t *samples_32 = data;
- int alloc_data_size= *data_size;
- int output_size;
+ int16_t *samples_16;
+ int32_t *samples_32;
+ int ret;
- *data_size=0;
+ *got_frame_ptr = 0;
if (s->max_framesize == 0) {
s->max_framesize =
@@ -586,15 +588,14 @@ static int flac_decode_frame(AVCodecContext *avctx,
}
bytes_read = (get_bits_count(&s->gb)+7)/8;
- /* check if allocated data size is large enough for output */
- output_size = s->blocksize * s->channels *
- av_get_bytes_per_sample(avctx->sample_fmt);
- if (output_size > alloc_data_size) {
- av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
- "allocated data size\n");
- return -1;
+ /* get output buffer */
+ s->frame.nb_samples = s->blocksize;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
- *data_size = output_size;
+ samples_16 = (int16_t *)s->frame.data[0];
+ samples_32 = (int32_t *)s->frame.data[0];
#define DECORRELATE(left, right)\
assert(s->channels == 2);\
@@ -639,6 +640,9 @@ static int flac_decode_frame(AVCodecContext *avctx,
buf_size - bytes_read, buf_size);
}
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
return bytes_read;
}
@@ -662,5 +666,6 @@ AVCodec ff_flac_decoder = {
.init = flac_decode_init,
.close = flac_decode_close,
.decode = flac_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};