# HG changeset patch # User lucabe # Date 1233845605 0 # Node ID 1b695f013cd3e9b2ea7de498b0910510d907f960 # Parent 7c628699cc3d1f70e80f8cba719814992ce20f75 Introduce a new num_frames field in RTPDemuxContext so that rtp_aac.c does not need to abuse read_buf_index diff -r 7c628699cc3d -r 1b695f013cd3 rtp.h --- a/rtp.h Thu Feb 05 08:59:38 2009 +0000 +++ b/rtp.h Thu Feb 05 14:53:25 2009 +0000 @@ -154,6 +154,7 @@ 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]; diff -r 7c628699cc3d -r 1b695f013cd3 rtp_aac.c --- a/rtp_aac.c Thu Feb 05 08:59:38 2009 +0000 +++ b/rtp_aac.c Thu Feb 05 14:53:25 2009 +0000 @@ -40,8 +40,8 @@ /* 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 @@ 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); diff -r 7c628699cc3d -r 1b695f013cd3 rtpenc.c --- a/rtpenc.c Thu Feb 05 08:59:38 2009 +0000 +++ b/rtpenc.c Thu Feb 05 14:53:25 2009 +0000 @@ -102,7 +102,7 @@ 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);