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Merge remote-tracking branch 'qatar/master'

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* qatar/master: (26 commits)
  adxenc: use AVCodec.encode2()
  adxenc: Use the AVFrame in ADXContext for coded_frame
  indeo4: fix out-of-bounds function call.
  configure: Restructure help output.
  configure: Internal-only components should not be command-line selectable.
  vorbisenc: use AVCodec.encode2()
  libvorbis: use AVCodec.encode2()
  libopencore-amrnbenc: use AVCodec.encode2()
  ra144enc: use AVCodec.encode2()
  nellymoserenc: use AVCodec.encode2()
  roqaudioenc: use AVCodec.encode2()
  libspeex: use AVCodec.encode2()
  libvo_amrwbenc: use AVCodec.encode2()
  libvo_aacenc: use AVCodec.encode2()
  wmaenc: use AVCodec.encode2()
  mpegaudioenc: use AVCodec.encode2()
  libmp3lame: use AVCodec.encode2()
  libgsmenc: use AVCodec.encode2()
  libfaac: use AVCodec.encode2()
  g726enc: use AVCodec.encode2()
  ...

Conflicts:
	configure
	libavcodec/Makefile
	libavcodec/ac3enc.c
	libavcodec/adxenc.c
	libavcodec/libgsm.c
	libavcodec/libvorbis.c
	libavcodec/vorbisenc.c
	libavcodec/wmaenc.c
	tests/ref/acodec/g722
	tests/ref/lavf/asf
	tests/ref/lavf/ffm
	tests/ref/lavf/mkv
	tests/ref/lavf/mpg
	tests/ref/lavf/rm
	tests/ref/lavf/ts
	tests/ref/seek/lavf_asf
	tests/ref/seek/lavf_ffm
	tests/ref/seek/lavf_mkv
	tests/ref/seek/lavf_mpg
	tests/ref/seek/lavf_rm
	tests/ref/seek/lavf_ts

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-03-21 23:47:44 +01:00
parent f1fdd208cc 6aba117f12
commit 967facb695
48 changed files with 1221 additions and 506 deletions
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48
libavcodec/wmaenc.c
View File

@ -20,6 +20,7 @@
*/
#include "avcodec.h"
#include "internal.h"
#include "wma.h"
#include "libavutil/avassert.h"
@ -87,7 +88,12 @@ static int encode_init(AVCodecContext * avctx){
avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate /
s->frame_len;
//av_log(NULL, AV_LOG_ERROR, "%d %d %d %d\n", s->block_align, avctx->bit_rate, s->frame_len, avctx->sample_rate);
avctx->frame_size= s->frame_len;
avctx->frame_size = avctx->delay = s->frame_len;
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = &s->frame;
avcodec_get_frame_defaults(avctx->coded_frame);
#endif
return 0;
}
@ -341,16 +347,17 @@ static int encode_frame(WMACodecContext *s, float (*src_coefs)[BLOCK_MAX_SIZE],
return put_bits_count(&s->pb)/8 - s->block_align;
}
static int encode_superframe(AVCodecContext *avctx,
unsigned char *buf, int buf_size, void *data){
static int encode_superframe(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
WMACodecContext *s = avctx->priv_data;
const short *samples = data;
int i, total_gain;
const int16_t *samples = (const int16_t *)frame->data[0];
int i, total_gain, ret;
s->block_len_bits= s->frame_len_bits; //required by non variable block len
s->block_len = 1 << s->block_len_bits;
apply_window_and_mdct(avctx, samples, avctx->frame_size);
apply_window_and_mdct(avctx, samples, frame->nb_samples);
if (s->ms_stereo) {
float a, b;
@ -364,24 +371,25 @@ static int encode_superframe(AVCodecContext *avctx,
}
}
if (buf_size < 2 * MAX_CODED_SUPERFRAME_SIZE) {
av_log(avctx, AV_LOG_ERROR, "output buffer size is too small\n");
return AVERROR(EINVAL);
if ((ret = ff_alloc_packet(avpkt, 2 * MAX_CODED_SUPERFRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
#if 1
total_gain= 128;
for(i=64; i; i>>=1){
int error= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
int error = encode_frame(s, s->coefs, avpkt->data, avpkt->size,
total_gain - i);
if(error<0)
total_gain-= i;
}
#else
total_gain= 90;
best= encode_frame(s, s->coefs, buf, buf_size, total_gain);
best = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
for(i=32; i; i>>=1){
int scoreL= encode_frame(s, s->coefs, buf, buf_size, total_gain-i);
int scoreR= encode_frame(s, s->coefs, buf, buf_size, total_gain+i);
int scoreL = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain - i);
int scoreR = encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain + i);
av_log(NULL, AV_LOG_ERROR, "%d %d %d (%d)\n", scoreL, best, scoreR, total_gain);
if(scoreL < FFMIN(best, scoreR)){
best = scoreL;
@ -393,7 +401,7 @@ static int encode_superframe(AVCodecContext *avctx,
}
#endif
encode_frame(s, s->coefs, buf, buf_size, total_gain);
encode_frame(s, s->coefs, avpkt->data, avpkt->size, total_gain);
av_assert0((put_bits_count(&s->pb) & 7) == 0);
i= s->block_align - (put_bits_count(&s->pb)+7)/8;
av_assert0(i>=0);
@ -402,7 +410,13 @@ static int encode_superframe(AVCodecContext *avctx,
flush_put_bits(&s->pb);
av_assert0(put_bits_ptr(&s->pb) - s->pb.buf == s->block_align);
return s->block_align;
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
avpkt->size = s->block_align;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_wmav1_encoder = {
@ -411,7 +425,7 @@ AVCodec ff_wmav1_encoder = {
.id = CODEC_ID_WMAV1,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode = encode_superframe,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
@ -423,7 +437,7 @@ AVCodec ff_wmav2_encoder = {
.id = CODEC_ID_WMAV2,
.priv_data_size = sizeof(WMACodecContext),
.init = encode_init,
.encode = encode_superframe,
.encode2 = encode_superframe,
.close = ff_wma_end,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),