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changing AVCodecContext codec -> *codec in AVStream so additions to AVCodecContext dont randomize AVStream and break binary compatibility

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Originally committed as revision 4453 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer
2005-07-17 22:24:36 +00:00
parent 4c1a012e9c
commit 01f4895c68
56 changed files with 1023 additions and 1024 deletions
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33
libavformat/rtp.c
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@@ -290,7 +290,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_t
return NULL;
}
} else {
switch(st->codec.codec_id) {
switch(st->codec->codec_id) {
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
case CODEC_ID_MP2:
@@ -307,7 +307,6 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_t
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
AVCodecContext codec;
int au_headers_length, au_header_size, i;
GetBitContext getbitcontext;
rtp_payload_data_t *infos;
@@ -317,8 +316,6 @@ static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
if (infos == NULL)
return -1;
codec = s->st->codec;
/* decode the first 2 bytes where are stored the AUHeader sections
length in bits */
au_headers_length = BE_16(buf);
@@ -407,7 +404,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
return -1;
#if defined(DEBUG) || 1
if (seq != ((s->seq + 1) & 0xffff)) {
av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
av_log(s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
}
#endif
@@ -428,7 +425,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
return 1;
}
} else {
switch(st->codec.codec_id) {
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
@@ -462,7 +459,7 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
break;
}
switch(st->codec.codec_id) {
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MPEG1VIDEO:
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
@@ -529,7 +526,7 @@ static int rtp_write_header(AVFormatContext *s1)
return -1;
st = s1->streams[0];
payload_type = rtp_get_payload_type(&st->codec);
payload_type = rtp_get_payload_type(st->codec);
if (payload_type < 0)
payload_type = RTP_PT_PRIVATE; /* private payload type */
s->payload_type = payload_type;
@@ -544,7 +541,7 @@ static int rtp_write_header(AVFormatContext *s1)
return AVERROR_IO;
s->max_payload_size = max_packet_size - 12;
switch(st->codec.codec_id) {
switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
@@ -663,7 +660,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1,
s->buf_ptr = s->buf + 4;
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
(s->cur_timestamp * 90000LL) / st->codec.sample_rate;
(s->cur_timestamp * 90000LL) / st->codec->sample_rate;
}
}
@@ -697,7 +694,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1,
memcpy(s->buf_ptr, buf1, size);
s->buf_ptr += size;
}
s->cur_timestamp += st->codec.frame_size;
s->cur_timestamp += st->codec->frame_size;
}
/* NOTE: a single frame must be passed with sequence header if
@@ -715,7 +712,7 @@ static void rtp_send_mpegvideo(AVFormatContext *s1,
while (size > 0) {
/* XXX: more correct headers */
h = 0;
if (st->codec.sub_id == 2)
if (st->codec->sub_id == 2)
h |= 1 << 26; /* mpeg 2 indicator */
q = s->buf;
*q++ = h >> 24;
@@ -723,7 +720,7 @@ static void rtp_send_mpegvideo(AVFormatContext *s1,
*q++ = h >> 8;
*q++ = h;
if (st->codec.sub_id == 2) {
if (st->codec->sub_id == 2) {
h = 0;
*q++ = h >> 24;
*q++ = h >> 16;
@@ -740,7 +737,7 @@ static void rtp_send_mpegvideo(AVFormatContext *s1,
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
av_rescale((int64_t)s->cur_timestamp * st->codec.time_base.num, 90000, st->codec.time_base.den); //FIXME pass timestamps
av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
rtp_send_data(s1, s->buf, q - s->buf);
buf1 += len;
@@ -765,7 +762,7 @@ static void rtp_send_raw(AVFormatContext *s1,
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
av_rescale((int64_t)s->cur_timestamp * st->codec.time_base.num, 90000, st->codec.time_base.den); //FIXME pass timestamps
av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
rtp_send_data(s1, buf1, len);
buf1 += len;
@@ -824,18 +821,18 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
s->first_packet = 0;
}
switch(st->codec.codec_id) {
switch(st->codec->codec_id) {
case CODEC_ID_PCM_MULAW:
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_U8:
case CODEC_ID_PCM_S8:
rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
break;
case CODEC_ID_PCM_U16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3: