generic-library/ffmpeg
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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/adpcmenc.c
View File

@@ -24,6 +24,7 @@
#include "bytestream.h"
#include "adpcm.h"
#include "adpcm_data.h"
#include "internal.h"
/**
* @file
@@ -144,8 +145,10 @@ static av_cold int adpcm_encode_init(AVCodecContext *avctx)
goto error;
}
#if FF_API_OLD_ENCODE_AUDIO
if (!(avctx->coded_frame = avcodec_alloc_frame()))
goto error;
#endif
return 0;
error:
@@ -156,7 +159,9 @@ error:
static av_cold int adpcm_encode_close(AVCodecContext *avctx)
{
ADPCMEncodeContext *s = avctx->priv_data;
#if FF_API_OLD_ENCODE_AUDIO
av_freep(&avctx->coded_frame);
#endif
av_freep(&s->paths);
av_freep(&s->node_buf);
av_freep(&s->nodep_buf);
@@ -473,23 +478,31 @@ static void adpcm_compress_trellis(AVCodecContext *avctx,
c->idelta = nodes[0]->step;
}
static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
int buf_size, void *data)
static int adpcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
int n, i, st;
int16_t *samples;
int n, i, st, pkt_size, ret;
const int16_t *samples;
uint8_t *dst;
ADPCMEncodeContext *c = avctx->priv_data;
uint8_t *buf;
dst = frame;
samples = data;
samples = (const int16_t *)frame->data[0];
st = avctx->channels == 2;
/* n = (BLKSIZE - 4 * avctx->channels) / (2 * 8 * avctx->channels); */
if (avctx->codec_id == CODEC_ID_ADPCM_SWF)
pkt_size = (2 + avctx->channels * (22 + 4 * (frame->nb_samples - 1)) + 7) / 8;
else
pkt_size = avctx->block_align;
if ((ret = ff_alloc_packet(avpkt, pkt_size))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
return ret;
}
dst = avpkt->data;
switch(avctx->codec->id) {
case CODEC_ID_ADPCM_IMA_WAV:
n = avctx->frame_size / 8;
n = frame->nb_samples / 8;
c->status[0].prev_sample = samples[0];
/* c->status[0].step_index = 0;
XXX: not sure how to init the state machine */
@@ -557,7 +570,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
{
int ch, i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size * 8);
init_put_bits(&pb, dst, pkt_size * 8);
for (ch = 0; ch < avctx->channels; ch++) {
put_bits(&pb, 9, (c->status[ch].prev_sample + 0x10000) >> 7);
@@ -581,16 +594,15 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
flush_put_bits(&pb);
dst += put_bits_count(&pb) >> 3;
break;
}
case CODEC_ID_ADPCM_SWF:
{
int i;
PutBitContext pb;
init_put_bits(&pb, dst, buf_size * 8);
init_put_bits(&pb, dst, pkt_size * 8);
n = avctx->frame_size - 1;
n = frame->nb_samples - 1;
// store AdpcmCodeSize
put_bits(&pb, 2, 2); // set 4-bit flash adpcm format
@@ -617,7 +629,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
av_free(buf);
} else {
for (i = 1; i < avctx->frame_size; i++) {
for (i = 1; i < frame->nb_samples; i++) {
put_bits(&pb, 4, adpcm_ima_compress_sample(&c->status[0],
samples[avctx->channels * i]));
if (avctx->channels == 2)
@@ -626,7 +638,6 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
}
flush_put_bits(&pb);
dst += put_bits_count(&pb) >> 3;
break;
}
case CODEC_ID_ADPCM_MS:
@@ -674,7 +685,7 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
}
break;
case CODEC_ID_ADPCM_YAMAHA:
n = avctx->frame_size / 2;
n = frame->nb_samples / 2;
if (avctx->trellis > 0) {
FF_ALLOC_OR_GOTO(avctx, buf, 2 * n * 2, error);
n *= 2;
@@ -700,7 +711,10 @@ static int adpcm_encode_frame(AVCodecContext *avctx, uint8_t *frame,
default:
return AVERROR(EINVAL);
}
return dst - frame;
avpkt->size = pkt_size;
*got_packet_ptr = 1;
return 0;
error:
return AVERROR(ENOMEM);
}
@@ -713,7 +727,7 @@ AVCodec ff_ ## name_ ## _encoder = { \
.id = id_, \
.priv_data_size = sizeof(ADPCMEncodeContext), \
.init = adpcm_encode_init, \
.encode = adpcm_encode_frame, \
.encode2 = adpcm_encode_frame, \
.close = adpcm_encode_close, \
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, \
AV_SAMPLE_FMT_NONE}, \