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Merge remote-tracking branch 'qatar/master'

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* qatar/master:
  avcodec_default_reget_buffer(): fix compilation in DEBUG mode
  fate: Overhaul WavPack coverage
  h264: fix mmxext chroma deblock to use correct TC values.
  flvdec: Remove the now redundant check for known broken metadata creator
  flvdec: Validate index entries added from metadata while reading
  rtsp: Handle requests from server to client
  movenc: use timestamps instead of frame_size for samples-per-packet
  movenc: use the first cluster duration as the tfhd default duration
  movenc: factorize calculation of cluster duration into a separate function
  doc/APIchanges: fill in missing dates and hashes.
  lavc: reorder AVCodecContext fields.
  lavc: reorder AVFrame fields.

Conflicts:
	doc/APIchanges
	libavcodec/avcodec.h
	libavformat/flvdec.c
	libavformat/movenc.c
	tests/fate/lossless-audio.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-02-28 03:38:58 +01:00
parent f1808e3048 d10319d87f
commit e3822886eb
9 changed files with 1204 additions and 1086 deletions
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58
libavformat/rtsp.c
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@@ -900,9 +900,13 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
char buf[4096], buf1[1024], *q;
unsigned char ch;
const char *p;
int ret, content_length, line_count = 0;
int ret, content_length, line_count = 0, request = 0;
unsigned char *content = NULL;
start:
line_count = 0;
request = 0;
content = NULL;
memset(reply, 0, sizeof(*reply));
/* parse reply (XXX: use buffers) */
@@ -938,9 +942,15 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
if (line_count == 0) {
/* get reply code */
get_word(buf1, sizeof(buf1), &p);
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
av_strlcpy(reply->reason, p, sizeof(reply->reason));
if (!strncmp(buf1, "RTSP/", 5)) {
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
av_strlcpy(reply->reason, p, sizeof(reply->reason));
} else {
av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
get_word(buf1, sizeof(buf1), &p); // object
request = 1;
}
} else {
ff_rtsp_parse_line(reply, p, rt, method);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
@@ -949,7 +959,7 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
line_count++;
}
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0' && !request)
av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
content_length = reply->content_length;
@@ -964,6 +974,44 @@ int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
else
av_free(content);
if (request) {
char buf[1024];
char base64buf[AV_BASE64_SIZE(sizeof(buf))];
const char* ptr = buf;
if (!strcmp(reply->reason, "OPTIONS")) {
snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
if (reply->seq)
av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
if (reply->session_id[0])
av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
reply->session_id);
} else {
snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
}
av_strlcat(buf, "\r\n", sizeof(buf));
if (rt->control_transport == RTSP_MODE_TUNNEL) {
av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
ptr = base64buf;
}
ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
rt->last_cmd_time = av_gettime();
/* Even if the request from the server had data, it is not the data
* that the caller wants or expects. The memory could also be leaked
* if the actual following reply has content data. */
if (content_ptr)
av_freep(content_ptr);
/* If method is set, this is called from ff_rtsp_send_cmd,
* where a reply to exactly this request is awaited. For
* callers from within packet reciving, we just want to
* return to the caller and go back to receiving packets. */
if (method)
goto start;
return 0;
}
if (rt->seq != reply->seq) {
av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
rt->seq, reply->seq);