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get rid of CODEC_ID_MPEG4AAC after next version bump, and change it to CODEC_ID_AAC where used

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Originally committed as revision 6954 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2006-11-09 11:37:30 +00:00
parent 6c4d16556b
commit cbee7a6944
7 changed files with 14 additions and 8 deletions
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2
libavcodec/avcodec.h
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@ -206,7 +206,9 @@ enum CodecID {
CODEC_ID_MP2= 0x15000,
CODEC_ID_MP3, /* prefered ID for MPEG Audio layer 1, 2 or3 decoding */
CODEC_ID_AAC,
#if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
CODEC_ID_MPEG4AAC,
#endif
CODEC_ID_AC3,
CODEC_ID_DTS,
CODEC_ID_VORBIS,

2
libavcodec/faad.c
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@ -326,7 +326,9 @@ AVCodec name ## _decoder = { \
// FIXME - raw AAC files - maybe just one entry will be enough
AAC_CODEC(CODEC_ID_AAC, aac);
#if LIBAVCODEC_VERSION_INT < ((52<<16)+(0<<8)+0)
// If it's mp4 file - usually embeded into Qt Mov
AAC_CODEC(CODEC_ID_MPEG4AAC, mpeg4aac);
#endif
#undef AAC_CODEC

3
libavformat/avidec.c
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@ -443,6 +443,9 @@ static int avi_read_header(AVFormatContext *s, AVFormatParameters *ap)
/* special case time: To support Xan DPCM, hardcode
* the format if Xxan is the video codec */
st->need_parsing = 1;
/* ADTS header is in extradata, AAC without header must be stored as exact frames, parser not needed and it will fail */
if (st->codec->codec_id == CODEC_ID_AAC && st->codec->extradata_size)
st->need_parsing = 0;
/* force parsing as several audio frames can be in
one packet */
if (xan_video)

5
libavformat/matroska.c
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@ -222,8 +222,7 @@ static CodecTags codec_tags[]={
{"A_AC3" , CODEC_ID_AC3},
{"A_DTS" , CODEC_ID_DTS},
{"A_VORBIS" , CODEC_ID_VORBIS},
{"A_AAC/MPEG2/" , CODEC_ID_AAC},
{"A_AAC/MPEG4/" , CODEC_ID_MPEG4AAC},
{"A_AAC/" , CODEC_ID_AAC},
{"A_WAVPACK4" , CODEC_ID_WAVPACK},
{NULL , CODEC_ID_NONE}
/* TODO: AC3-9/10 (?), Real, Musepack, Quicktime */
@ -2246,7 +2245,7 @@ matroska_read_header (AVFormatContext *s,
}
if (codec_id==CODEC_ID_AAC || codec_id==CODEC_ID_MPEG4AAC) {
if (codec_id==CODEC_ID_AAC) {
MatroskaAudioTrack *audiotrack = (MatroskaAudioTrack *) track;
int profile = matroska_aac_profile(track->codec_id);
int sri = matroska_aac_sri(audiotrack->internal_samplerate);

2
libavformat/riff.c
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@ -186,7 +186,7 @@ const CodecTag codec_wav_tags[] = {
{ CODEC_ID_WMAV1, 0x160 },
{ CODEC_ID_WMAV2, 0x161 },
{ CODEC_ID_AAC, 0x706d },
{ CODEC_ID_MPEG4AAC, 0xff },
{ CODEC_ID_AAC, 0xff },
{ CODEC_ID_VORBIS, ('V'<<8)+'o' }, //HACK/FIXME, does vorbis in WAV/AVI have an (in)official id?
{ CODEC_ID_SONIC, 0x2048 },
{ CODEC_ID_SONIC_LS, 0x2048 },

6
libavformat/rtp.c
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@ -186,7 +186,7 @@ AVRtpPayloadType_t AVRtpPayloadTypes[]=
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_MPEG4AAC};
static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
@ -564,7 +564,7 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestam
pkt->pts = addend + delta_timestamp;
}
break;
case CODEC_ID_MPEG4AAC:
case CODEC_ID_AAC:
case CODEC_ID_H264:
case CODEC_ID_MPEG4:
pkt->pts = timestamp;
@ -694,7 +694,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
// moved from below, verbatim. this is because this section handles packets, and the lower switch handles
// timestamps.
// TODO: Put this into a dynamic packet handler...
case CODEC_ID_MPEG4AAC:
case CODEC_ID_AAC:
if (rtp_parse_mp4_au(s, buf))
return -1;
{

2
libavformat/rtsp.c
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@ -251,7 +251,7 @@ static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value
{
switch (codec->codec_id) {
case CODEC_ID_MPEG4:
case CODEC_ID_MPEG4AAC:
case CODEC_ID_AAC:
if (!strcmp(attr, "config")) {
/* decode the hexa encoded parameter */
int len = hex_to_data(NULL, value);