virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-21 10:55:51 +02:00
Code Issues Releases Activity

aacenc: use planar sample format

Browse Source
This commit is contained in:
Justin Ruggles 2012-08-25 13:04:33 -04:00
parent 095be4fbf6
commit f3e2d68df6
1 changed files with 15 additions and 18 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

33
libavcodec/aacenc.c
View File

@ -479,31 +479,28 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
}
/*
* Deinterleave input samples.
* Copy input samples.
* Channels are reordered from Libav's default order to AAC order.
*/
static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame)
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
{
int ch, i;
const int sinc = s->channels;
const uint8_t *channel_map = aac_chan_maps[sinc - 1];
int ch;
int end = 2048 + (frame ? frame->nb_samples : 0);
const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
/* deinterleave and remap input samples */
for (ch = 0; ch < sinc; ch++) {
/* copy and remap input samples */
for (ch = 0; ch < s->channels; ch++) {
/* copy last 1024 samples of previous frame to the start of the current frame */
memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
/* deinterleave */
i = 2048;
/* copy new samples and zero any remaining samples */
if (frame) {
const float *sptr = ((const float *)frame->data[0]) + channel_map[ch];
for (; i < 2048 + frame->nb_samples; i++) {
s->planar_samples[ch][i] = *sptr;
sptr += sinc;
}
memcpy(&s->planar_samples[ch][2048],
frame->extended_data[channel_map[ch]],
frame->nb_samples * sizeof(s->planar_samples[0][0]));
}
memset(&s->planar_samples[ch][i], 0,
(3072 - i) * sizeof(s->planar_samples[0][0]));
memset(&s->planar_samples[ch][end], 0,
(3072 - end) * sizeof(s->planar_samples[0][0]));
}
}
@ -526,7 +523,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
return ret;
}
deinterleave_input_samples(s, frame);
copy_input_samples(s, frame);
if (s->psypp)
ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
@ -826,7 +823,7 @@ AVCodec ff_aac_encoder = {
.close = aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
CODEC_CAP_EXPERIMENTAL,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
.priv_class = &aacenc_class,