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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/cook.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/cook.c')
-rw-r--r--libavcodec/cook.c43
1 files changed, 30 insertions, 13 deletions
diff --git a/libavcodec/cook.c b/libavcodec/cook.c
index 8b0a351495..81a1aae9d1 100644
--- a/libavcodec/cook.c
+++ b/libavcodec/cook.c
@@ -122,6 +122,7 @@ typedef struct cook {
void (* saturate_output) (struct cook *q, int chan, float *out);
AVCodecContext* avctx;
+ AVFrame frame;
GetBitContext gb;
/* stream data */
int nb_channels;
@@ -131,6 +132,7 @@ typedef struct cook {
int samples_per_channel;
/* states */
AVLFG random_state;
+ int discarded_packets;
/* transform data */
FFTContext mdct_ctx;
@@ -896,7 +898,8 @@ mlt_compensate_output(COOKContext *q, float *decode_buffer,
float *out, int chan)
{
imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
- q->saturate_output (q, chan, out);
+ if (out)
+ q->saturate_output(q, chan, out);
}
@@ -953,24 +956,28 @@ static void decode_subpacket(COOKContext *q, COOKSubpacket *p,
* @param avctx pointer to the AVCodecContext
*/
-static int cook_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt) {
+static int cook_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data;
- int i, out_size;
+ float *samples = NULL;
+ int i, ret;
int offset = 0;
int chidx = 0;
if (buf_size < avctx->block_align)
return buf_size;
- out_size = q->nb_channels * q->samples_per_channel *
- 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 */
+ if (q->discarded_packets >= 2) {
+ q->frame.nb_samples = q->samples_per_channel;
+ if ((ret = avctx->get_buffer(avctx, &q->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ samples = (float *)q->frame.data[0];
}
/* estimate subpacket sizes */
@@ -990,15 +997,21 @@ static int cook_decode_frame(AVCodecContext *avctx,
q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
q->subpacket[i].ch_idx = chidx;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
- decode_subpacket(q, &q->subpacket[i], buf + offset, data);
+ decode_subpacket(q, &q->subpacket[i], buf + offset, samples);
offset += q->subpacket[i].size;
chidx += q->subpacket[i].num_channels;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
}
- *data_size = out_size;
/* Discard the first two frames: no valid audio. */
- if (avctx->frame_number < 2) *data_size = 0;
+ if (q->discarded_packets < 2) {
+ q->discarded_packets++;
+ *got_frame_ptr = 0;
+ return avctx->block_align;
+ }
+
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = q->frame;
return avctx->block_align;
}
@@ -1246,6 +1259,9 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)
else
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
+ avcodec_get_frame_defaults(&q->frame);
+ avctx->coded_frame = &q->frame;
+
#ifdef DEBUG
dump_cook_context(q);
#endif
@@ -1262,5 +1278,6 @@ AVCodec ff_cook_decoder =
.init = cook_decode_init,
.close = cook_decode_close,
.decode = cook_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("COOK"),
};