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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
Download Patch File Download Diff File Expand all files Collapse all files

69
libavcodec/libvorbis.c
View File

@@ -29,9 +29,11 @@
#include "libavutil/fifo.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "audio_frame_queue.h"
#include "bytestream.h"
#include "internal.h"
#include "vorbis.h"
#include "vorbis_parser.h"
#undef NDEBUG
#include <assert.h>
@@ -56,6 +58,8 @@ typedef struct OggVorbisContext {
vorbis_comment vc; /**< VorbisComment info */
ogg_packet op; /**< ogg packet */
double iblock; /**< impulse block bias option */
VorbisParseContext vp; /**< parse context to get durations */
AudioFrameQueue afq; /**< frame queue for timestamps */
} OggVorbisContext;
static const AVOption options[] = {
@@ -186,7 +190,10 @@ static av_cold int oggvorbis_encode_close(AVCodecContext *avctx)
vorbis_info_clear(&s->vi);
av_fifo_free(s->pkt_fifo);
ff_af_queue_close(&s->afq);
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&avctx->extradata);
return 0;
@@ -247,9 +254,15 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
offset += header_code.bytes;
assert(offset == avctx->extradata_size);
if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
return ret;
}
vorbis_comment_clear(&s->vc);
avctx->frame_size = OGGVORBIS_FRAME_SIZE;
ff_af_queue_init(avctx, &s->afq);
s->pkt_fifo = av_fifo_alloc(BUFFER_SIZE);
if (!s->pkt_fifo) {
@@ -257,11 +270,13 @@ static av_cold int oggvorbis_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame) {
ret = AVERROR(ENOMEM);
goto error;
}
#endif
return 0;
error:
@@ -269,17 +284,17 @@ error:
return ret;
}
static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
int buf_size, void *data)
static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
OggVorbisContext *s = avctx->priv_data;
ogg_packet op;
float *audio = data;
int pkt_size, ret;
int ret, duration;
/* send samples to libvorbis */
if (data) {
const int samples = avctx->frame_size;
if (frame) {
const float *audio = (const float *)frame->data[0];
const int samples = frame->nb_samples;
float **buffer;
int c, channels = s->vi.channels;
@@ -295,6 +310,8 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
return vorbis_error_to_averror(ret);
}
if ((ret = ff_af_queue_add(&s->afq, frame) < 0))
return ret;
} else {
if (!s->eof)
if ((ret = vorbis_analysis_wrote(&s->vd, 0)) < 0) {
@@ -330,22 +347,34 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, unsigned char *packets,
return vorbis_error_to_averror(ret);
}
/* output then next packet from the output buffer, if available */
pkt_size = 0;
if (av_fifo_size(s->pkt_fifo) >= sizeof(ogg_packet)) {
av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
pkt_size = op.bytes;
// FIXME: we should use the user-supplied pts and duration
avctx->coded_frame->pts = ff_samples_to_time_base(avctx,
op.granulepos);
if (pkt_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* check for available packets */
if (av_fifo_size(s->pkt_fifo) < sizeof(ogg_packet))
return 0;
av_fifo_generic_read(s->pkt_fifo, &op, sizeof(ogg_packet), NULL);
if ((ret = ff_alloc_packet(avpkt, op.bytes))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
av_fifo_generic_read(s->pkt_fifo, avpkt->data, op.bytes, NULL);
avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
if (duration > 0) {
/* we do not know encoder delay until we get the first packet from
* libvorbis, so we have to update the AudioFrameQueue counts */
if (!avctx->delay) {
avctx->delay = duration;
s->afq.remaining_delay += duration;
s->afq.remaining_samples += duration;
}
av_fifo_generic_read(s->pkt_fifo, packets, pkt_size, NULL);
ff_af_queue_remove(&s->afq, duration, &avpkt->pts, &avpkt->duration);
}
return pkt_size;
*got_packet_ptr = 1;
return 0;
}
AVCodec ff_libvorbis_encoder = {
@@ -354,7 +383,7 @@ AVCodec ff_libvorbis_encoder = {
.id = CODEC_ID_VORBIS,
.priv_data_size = sizeof(OggVorbisContext),
.init = oggvorbis_encode_init,
.encode = oggvorbis_encode_frame,
.encode2 = oggvorbis_encode_frame,
.close = oggvorbis_encode_close,
.capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,