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(e)ac3enc: use planar sample format

This commit is contained in:
Justin Ruggles 2012-08-25 13:11:10 -04:00
parent f3e2d68df6
commit b1540fc884
4 changed files with 12 additions and 20 deletions

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@ -149,7 +149,7 @@ AVCodec ff_ac3_fixed_encoder = {
.init = ac3_fixed_encode_init,
.encode2 = ff_ac3_fixed_encode_frame,
.close = ff_ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.priv_class = &ac3enc_class,

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@ -148,7 +148,7 @@ AVCodec ff_ac3_encoder = {
.init = ff_ac3_encode_init,
.encode2 = ff_ac3_float_encode_frame,
.close = ff_ac3_encode_close,
.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("ATSC A/52A (AC-3)"),
.priv_class = &ac3enc_class,

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@ -65,30 +65,23 @@ alloc_fail:
/*
* Deinterleave input samples.
* Copy input samples.
* Channels are reordered from Libav's default order to AC-3 order.
*/
static void deinterleave_input_samples(AC3EncodeContext *s,
const SampleType *samples)
static void copy_input_samples(AC3EncodeContext *s, SampleType **samples)
{
int ch, i;
int ch;
/* deinterleave and remap input samples */
/* copy and remap input samples */
for (ch = 0; ch < s->channels; ch++) {
const SampleType *sptr;
int sinc;
/* copy last 256 samples of previous frame to the start of the current frame */
memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][AC3_BLOCK_SIZE * s->num_blocks],
AC3_BLOCK_SIZE * sizeof(s->planar_samples[0][0]));
/* deinterleave */
sinc = s->channels;
sptr = samples + s->channel_map[ch];
for (i = AC3_BLOCK_SIZE; i < AC3_BLOCK_SIZE * (s->num_blocks + 1); i++) {
s->planar_samples[ch][i] = *sptr;
sptr += sinc;
}
/* copy new samples for current frame */
memcpy(&s->planar_samples[ch][AC3_BLOCK_SIZE],
samples[s->channel_map[ch]],
AC3_BLOCK_SIZE * s->num_blocks * sizeof(s->planar_samples[0][0]));
}
}
@ -400,7 +393,6 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
AC3EncodeContext *s = avctx->priv_data;
const SampleType *samples = (const SampleType *)frame->data[0];
int ret;
if (s->options.allow_per_frame_metadata) {
@ -412,7 +404,7 @@ int AC3_NAME(encode_frame)(AVCodecContext *avctx, AVPacket *avpkt,
if (s->bit_alloc.sr_code == 1 || s->eac3)
ff_ac3_adjust_frame_size(s);
deinterleave_input_samples(s, samples);
copy_input_samples(s, (SampleType **)frame->extended_data);
apply_mdct(s);

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@ -254,7 +254,7 @@ AVCodec ff_eac3_encoder = {
.init = ff_ac3_encode_init,
.encode2 = ff_ac3_float_encode_frame,
.close = ff_ac3_encode_close,
.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("ATSC A/52 E-AC-3"),
.priv_class = &eac3enc_class,