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Introduce a new num_frames field in RTPDemuxContext so that rtp_aac.c

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does not need to abuse read_buf_index

Originally committed as revision 17004 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Luca Abeni 2009-02-05 14:53:25 +00:00
parent 468d298d0c
commit 21da81d784
3 changed files with 7 additions and 6 deletions
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1
libavformat/rtp.h
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@ -154,6 +154,7 @@ struct RTPDemuxContext {
struct MpegTSContext *ts; /* only used for MP2T payloads */
int read_buf_index;
int read_buf_size;
int num_frames;
/* used to send back RTCP RR */
URLContext *rtp_ctx;
char hostname[256];

10
libavformat/rtp_aac.c
View File

@ -40,8 +40,8 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
/* test if the packet must be sent */
len = (s->buf_ptr - s->buf);
if ((s->read_buf_index == MAX_FRAMES_PER_PACKET) || (len && (len + size) > max_packet_size)) {
int au_size = s->read_buf_index * 2;
if ((s->num_frames == MAX_FRAMES_PER_PACKET) || (len && (len + size) > max_packet_size)) {
int au_size = s->num_frames * 2;
p = s->buf + MAX_AU_HEADERS_SIZE - au_size - 2;
if (p != s->buf) {
@ -53,15 +53,15 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
ff_rtp_send_data(s1, p, s->buf_ptr - p, 1);
s->read_buf_index = 0;
s->num_frames = 0;
}
if (s->read_buf_index == 0) {
if (s->num_frames == 0) {
s->buf_ptr = s->buf + MAX_AU_HEADERS_SIZE;
s->timestamp = s->cur_timestamp;
}
if (size < max_packet_size) {
p = s->buf + s->read_buf_index++ * 2 + 2;
p = s->buf + s->num_frames++ * 2 + 2;
*p++ = size >> 5;
*p = (size & 0x1F) << 3;
memcpy(s->buf_ptr, buff, size);

2
libavformat/rtpenc.c
View File

@ -102,7 +102,7 @@ static int rtp_write_header(AVFormatContext *s1)
s->buf_ptr = s->buf;
break;
case CODEC_ID_AAC:
s->read_buf_index = 0;
s->num_frames = 0;
default:
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);