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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2011-12-03 02:08:55 +01:00
parent 12804348f5 d268b79e34
commit e4de71677f
87 changed files with 2226 additions and 1307 deletions
Download Patch File Download Diff File Expand all files Collapse all files

89
libavcodec/mpegaudiodec.c
View File

@@ -79,6 +79,7 @@ typedef struct MPADecodeContext {
int err_recognition;
AVCodecContext* avctx;
MPADSPContext mpadsp;
AVFrame frame;
} MPADecodeContext;
#if CONFIG_FLOAT
@@ -479,6 +480,10 @@ static av_cold int decode_init(AVCodecContext * avctx)
if (avctx->codec_id == CODEC_ID_MP3ADU)
s->adu_mode = 1;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
@@ -1581,7 +1586,7 @@ static int mp_decode_layer3(MPADecodeContext *s)
static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
const uint8_t *buf, int buf_size)
{
int i, nb_frames, ch;
int i, nb_frames, ch, ret;
OUT_INT *samples_ptr;
init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
@@ -1629,8 +1634,16 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
assert(i <= buf_size - HEADER_SIZE && i >= 0);
memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
s->last_buf_size += i;
}
break;
/* get output buffer */
if (!samples) {
s->frame.nb_samples = s->avctx->frame_size;
if ((ret = s->avctx->get_buffer(s->avctx, &s->frame)) < 0) {
av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (OUT_INT *)s->frame.data[0];
}
/* apply the synthesis filter */
@@ -1650,7 +1663,7 @@ static int mp_decode_frame(MPADecodeContext *s, OUT_INT *samples,
return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
}
static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
@@ -1658,7 +1671,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int out_size;
OUT_INT *out_samples = data;
if (buf_size < HEADER_SIZE)
return AVERROR_INVALIDDATA;
@@ -1681,10 +1693,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT))
return AVERROR(EINVAL);
*data_size = 0;
if (s->frame_size <= 0 || s->frame_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
return AVERROR_INVALIDDATA;
@@ -1693,9 +1701,10 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
buf_size= s->frame_size;
}
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
out_size = mp_decode_frame(s, NULL, buf, buf_size);
if (out_size >= 0) {
*data_size = out_size;
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
avctx->sample_rate = s->sample_rate;
//FIXME maybe move the other codec info stuff from above here too
} else {
@@ -1704,6 +1713,7 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *data_size,
If there is more data in the packet, just consume the bad frame
instead of returning an error, which would discard the whole
packet. */
*got_frame_ptr = 0;
if (buf_size == avpkt->size)
return out_size;
}
@@ -1719,15 +1729,14 @@ static void flush(AVCodecContext *avctx)
}
#if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
static int decode_frame_adu(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int len, out_size;
OUT_INT *out_samples = data;
len = buf_size;
@@ -1757,9 +1766,6 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
if (*data_size < avctx->frame_size * avctx->channels * sizeof(OUT_INT))
return AVERROR(EINVAL);
s->frame_size = len;
#if FF_API_PARSE_FRAME
@@ -1767,9 +1773,11 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
out_size = buf_size;
else
#endif
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
out_size = mp_decode_frame(s, NULL, buf, buf_size);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
*data_size = out_size;
return buf_size;
}
#endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
@@ -1780,6 +1788,7 @@ static int decode_frame_adu(AVCodecContext *avctx, void *data, int *data_size,
* Context for MP3On4 decoder
*/
typedef struct MP3On4DecodeContext {
AVFrame *frame;
int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
int syncword; ///< syncword patch
const uint8_t *coff; ///< channel offsets in output buffer
@@ -1843,7 +1852,8 @@ static int decode_init_mp3on4(AVCodecContext * avctx)
return AVERROR_INVALIDDATA;
}
avpriv_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
avctx->extradata_size * 8, 1);
if (!cfg.chan_config || cfg.chan_config > 7) {
av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
return AVERROR_INVALIDDATA;
@@ -1870,6 +1880,7 @@ static int decode_init_mp3on4(AVCodecContext * avctx)
// Put decoder context in place to make init_decode() happy
avctx->priv_data = s->mp3decctx[0];
decode_init(avctx);
s->frame = avctx->coded_frame;
// Restore mp3on4 context pointer
avctx->priv_data = s;
s->mp3decctx[0]->adu_mode = 1; // Set adu mode
@@ -1914,9 +1925,8 @@ static void flush_mp3on4(AVCodecContext *avctx)
}
static int decode_frame_mp3on4(AVCodecContext * avctx,
void *data, int *data_size,
AVPacket *avpkt)
static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
@@ -1924,14 +1934,17 @@ static int decode_frame_mp3on4(AVCodecContext * avctx,
MPADecodeContext *m;
int fsize, len = buf_size, out_size = 0;
uint32_t header;
OUT_INT *out_samples = data;
OUT_INT *out_samples;
OUT_INT *outptr, *bp;
int fr, j, n, ch;
int fr, j, n, ch, ret;
if (*data_size < MPA_FRAME_SIZE * avctx->channels * sizeof(OUT_INT)) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
/* get output buffer */
s->frame->nb_samples = MPA_FRAME_SIZE;
if ((ret = avctx->get_buffer(avctx, s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_samples = (OUT_INT *)s->frame->data[0];
// Discard too short frames
if (buf_size < HEADER_SIZE)
@@ -1990,7 +2003,10 @@ static int decode_frame_mp3on4(AVCodecContext * avctx,
/* update codec info */
avctx->sample_rate = s->mp3decctx[0]->sample_rate;
*data_size = out_size;
s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
*got_frame_ptr = 1;
*(AVFrame *)data = *s->frame;
return buf_size;
}
#endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
@@ -2005,7 +2021,9 @@ AVCodec ff_mp1_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
@@ -2020,7 +2038,9 @@ AVCodec ff_mp2_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
@@ -2035,7 +2055,9 @@ AVCodec ff_mp3_decoder = {
.init = decode_init,
.decode = decode_frame,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
@@ -2050,7 +2072,9 @@ AVCodec ff_mp3adu_decoder = {
.init = decode_init,
.decode = decode_frame_adu,
#if FF_API_PARSE_FRAME
.capabilities = CODEC_CAP_PARSE_ONLY,
.capabilities = CODEC_CAP_PARSE_ONLY | CODEC_CAP_DR1,
#else
.capabilities = CODEC_CAP_DR1,
#endif
.flush = flush,
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
@@ -2065,6 +2089,7 @@ AVCodec ff_mp3on4_decoder = {
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = CODEC_CAP_DR1,
.flush = flush_mp3on4,
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
};