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Unified Diff: patched-ffmpeg-mt/libavformat/rtsp.c

Issue 789004: ffmpeg roll of source to mar 9 version... (Closed) Base URL: svn://chrome-svn/chrome/trunk/deps/third_party/ffmpeg/
Patch Set: '' Created 10 years, 9 months ago
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« patched-ffmpeg-mt/libavcodec/mpeg4video_es_bsf.c ('K') | « patched-ffmpeg-mt/libavformat/rtsp.h ('k') | patched-ffmpeg-mt/libavformat/rtspenc.c » ('j') | no next file with comments »
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Index: patched-ffmpeg-mt/libavformat/rtsp.c
===================================================================
--- patched-ffmpeg-mt/libavformat/rtsp.c (revision 41250)
+++ patched-ffmpeg-mt/libavformat/rtsp.c (working copy)
@@ -19,9 +19,6 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/* needed by inet_aton() */
-#define _SVID_SOURCE
-
#include "libavutil/base64.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
@@ -37,8 +34,8 @@
#include "rtpdec.h"
#include "rdt.h"
-#include "rtp_asf.h"
-#include "rtp_vorbis.h"
+#include "rtpdec_asf.h"
+#include "rtpdec_vorbis.h"
//#define DEBUG
//#define DEBUG_RTP_TCP
@@ -131,42 +128,38 @@
if (c && c->name)
c_name = c->name;
else
- c_name = (char *) NULL;
+ c_name = "(null)";
- if (c_name) {
- get_word_sep(buf, sizeof(buf), "/", &p);
- i = atoi(buf);
- switch (codec->codec_type) {
- case CODEC_TYPE_AUDIO:
- av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
- codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
- codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
- if (i > 0) {
- codec->sample_rate = i;
- get_word_sep(buf, sizeof(buf), "/", &p);
- i = atoi(buf);
- if (i > 0)
- codec->channels = i;
- // TODO: there is a bug here; if it is a mono stream, and
- // less than 22000Hz, faad upconverts to stereo and twice
- // the frequency. No problem, but the sample rate is being
- // set here by the sdp line. Patch on its way. (rdm)
- }
- av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
- codec->sample_rate);
- av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
- codec->channels);
- break;
- case CODEC_TYPE_VIDEO:
- av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
- break;
- default:
- break;
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ switch (codec->codec_type) {
+ case CODEC_TYPE_AUDIO:
+ av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
+ codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
+ codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
+ if (i > 0) {
+ codec->sample_rate = i;
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ if (i > 0)
+ codec->channels = i;
+ // TODO: there is a bug here; if it is a mono stream, and
+ // less than 22000Hz, faad upconverts to stereo and twice
+ // the frequency. No problem, but the sample rate is being
+ // set here by the sdp line. Patch on its way. (rdm)
}
- return 0;
+ av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
+ codec->sample_rate);
+ av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
+ codec->channels);
+ break;
+ case CODEC_TYPE_VIDEO:
+ av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
+ break;
+ default:
+ break;
}
-
- return -1;
+ return 0;
}
/* return the length and optionally the data */
@@ -254,8 +247,8 @@
/* parse the attribute line from the fmtp a line of an sdp resonse. This
* is broken out as a function because it is used in rtp_h264.c, which is
* forthcoming. */
-int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
- char *value, int value_size)
+int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
+ char *value, int value_size)
{
skip_spaces(p);
if (**p) {
@@ -283,8 +276,8 @@
RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;
/* loop on each attribute */
- while (rtsp_next_attr_and_value(&p, attr, sizeof(attr),
- value, sizeof(value))) {
+ while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr),
+ value, sizeof(value))) {
/* grab the codec extra_data from the config parameter of the fmtp
* line */
sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
@@ -363,7 +356,7 @@
if (strcmp(buf1, "IP4") != 0)
return;
get_word_sep(buf1, sizeof(buf1), "/", &p);
- if (inet_aton(buf1, &sdp_ip) == 0)
+ if (ff_inet_aton(buf1, &sdp_ip) == 0)
return;
ttl = 16;
if (*p == '/') {
@@ -453,8 +446,8 @@
rtsp_st = st->priv_data;
/* XXX: may need to add full url resolution */
- url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
- NULL, NULL, 0, p);
+ ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
+ NULL, NULL, 0, p);
if (proto[0] == '\0') {
/* relative control URL */
if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
@@ -576,8 +569,9 @@
}
/* close and free RTSP streams */
-static void rtsp_close_streams(RTSPState *rt)
+void ff_rtsp_close_streams(AVFormatContext *s)
{
+ RTSPState *rt = s->priv_data;
int i;
RTSPStream *rtsp_st;
@@ -585,7 +579,15 @@
rtsp_st = rt->rtsp_streams[i];
if (rtsp_st) {
if (rtsp_st->transport_priv) {
- if (rt->transport == RTSP_TRANSPORT_RDT)
+ if (s->oformat) {
+ AVFormatContext *rtpctx = rtsp_st->transport_priv;
+ av_write_trailer(rtpctx);
+ url_fclose(rtpctx->pb);
+ av_metadata_free(&rtpctx->streams[0]->metadata);
+ av_metadata_free(&rtpctx->metadata);
+ av_free(rtpctx->streams[0]);
+ av_free(rtpctx);
+ } else if (rt->transport == RTSP_TRANSPORT_RDT)
ff_rdt_parse_close(rtsp_st->transport_priv);
else
rtp_parse_close(rtsp_st->transport_priv);
@@ -605,6 +607,50 @@
av_freep(&rt->auth_b64);
}
+static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st, URLContext *handle) {
+ AVFormatContext *rtpctx;
+ int ret;
+ AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
+
+ if (!rtp_format)
+ return NULL;
+
+ /* Allocate an AVFormatContext for each output stream */
+ rtpctx = avformat_alloc_context();
+ if (!rtpctx)
+ return NULL;
+
+ rtpctx->oformat = rtp_format;
+ if (!av_new_stream(rtpctx, 0)) {
+ av_free(rtpctx);
+ return NULL;
+ }
+ /* Copy the max delay setting; the rtp muxer reads this. */
+ rtpctx->max_delay = s->max_delay;
+ /* Copy other stream parameters. */
+ rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
+
+ /* Remove the local codec, link to the original codec
+ * context instead, to give the rtp muxer access to
+ * codec parameters. */
+ av_free(rtpctx->streams[0]->codec);
+ rtpctx->streams[0]->codec = st->codec;
+
+ url_fdopen(&rtpctx->pb, handle);
+ ret = av_write_header(rtpctx);
+
+ if (ret) {
+ url_fclose(rtpctx->pb);
+ av_free(rtpctx->streams[0]);
+ av_free(rtpctx);
+ return NULL;
+ }
+
+ /* Copy the RTP AVStream timebase back to the original AVStream */
+ st->time_base = rtpctx->streams[0]->time_base;
+ return rtpctx;
+}
+
static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
{
RTSPState *rt = s->priv_data;
@@ -616,7 +662,11 @@
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- if (rt->transport == RTSP_TRANSPORT_RDT)
+ if (s->oformat) {
+ rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
+ /* Ownage of rtp_handle is passed to the rtp mux context */
+ rtsp_st->rtp_handle = NULL;
+ } else if (rt->transport == RTSP_TRANSPORT_RDT)
rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
rtsp_st->dynamic_protocol_context,
rtsp_st->dynamic_handler);
@@ -638,7 +688,7 @@
return 0;
}
-#if CONFIG_RTSP_DEMUXER
+#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
static int rtsp_probe(AVProbeData *p)
{
if (av_strstart(p->filename, "rtsp:", NULL))
@@ -750,7 +800,7 @@
if (*p == '=') {
p++;
get_word_sep(buf, sizeof(buf), ";,", &p);
- if (inet_aton(buf, &ipaddr))
+ if (ff_inet_aton(buf, &ipaddr))
th->destination = ntohl(ipaddr.s_addr);
}
}
@@ -766,7 +816,7 @@
}
}
-void rtsp_parse_line(RTSPMessageHeader *reply, const char *buf)
+void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf)
{
const char *p;
@@ -828,30 +878,9 @@
}
}
-/**
- * Read a RTSP message from the server, or prepare to read data
- * packets if we're reading data interleaved over the TCP/RTSP
- * connection as well.
- *
- * @param s RTSP demuxer context
- * @param reply pointer where the RTSP message header will be stored
- * @param content_ptr pointer where the RTSP message body, if any, will
- * be stored (length is in reply)
- * @param return_on_interleaved_data whether the function may return if we
- * encounter a data marker ('$'), which precedes data
- * packets over interleaved TCP/RTSP connections. If this
- * is set, this function will return 1 after encountering
- * a '$'. If it is not set, the function will skip any
- * data packets (if they are encountered), until a reply
- * has been fully parsed. If no more data is available
- * without parsing a reply, it will return an error.
- *
- * @returns 1 if a data packets is ready to be received, -1 on error,
- * and 0 on success.
- */
-static int rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
- unsigned char **content_ptr,
- int return_on_interleaved_data)
+int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ int return_on_interleaved_data)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024], *q;
@@ -900,7 +929,7 @@
get_word(buf1, sizeof(buf1), &p);
reply->status_code = atoi(buf1);
} else {
- rtsp_parse_line(reply, p);
+ ff_rtsp_parse_line(reply, p);
av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
}
@@ -936,7 +965,10 @@
return 0;
}
-static void rtsp_send_cmd_async(AVFormatContext *s, const char *cmd)
+void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
+ const char *cmd,
+ const unsigned char *send_content,
+ int send_content_length)
{
RTSPState *rt = s->priv_data;
char buf[4096], buf1[1024];
@@ -953,23 +985,44 @@
av_strlcatf(buf, sizeof(buf),
"Authorization: Basic %s\r\n",
rt->auth_b64);
+ if (send_content_length > 0 && send_content)
+ av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
av_strlcat(buf, "\r\n", sizeof(buf));
dprintf(s, "Sending:\n%s--\n", buf);
url_write(rt->rtsp_hd, buf, strlen(buf));
+ if (send_content_length > 0 && send_content)
+ url_write(rt->rtsp_hd, send_content, send_content_length);
rt->last_cmd_time = av_gettime();
}
-static void rtsp_send_cmd(AVFormatContext *s,
- const char *cmd, RTSPMessageHeader *reply,
- unsigned char **content_ptr)
+void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *cmd)
{
- rtsp_send_cmd_async(s, cmd);
+ ff_rtsp_send_cmd_with_content_async(s, cmd, NULL, 0);
+}
- rtsp_read_reply(s, reply, content_ptr, 0);
+void ff_rtsp_send_cmd(AVFormatContext *s,
+ const char *cmd, RTSPMessageHeader *reply,
+ unsigned char **content_ptr)
+{
+ ff_rtsp_send_cmd_async(s, cmd);
+
+ ff_rtsp_read_reply(s, reply, content_ptr, 0);
}
+void ff_rtsp_send_cmd_with_content(AVFormatContext *s,
+ const char *cmd,
+ RTSPMessageHeader *reply,
+ unsigned char **content_ptr,
+ const unsigned char *send_content,
+ int send_content_length)
+{
+ ff_rtsp_send_cmd_with_content_async(s, cmd, send_content, send_content_length);
+
+ ff_rtsp_read_reply(s, reply, content_ptr, 0);
+}
+
/**
* @returns 0 on success, <0 on error, 1 if protocol is unavailable.
*/
@@ -1034,8 +1087,8 @@
/* first try in specified port range */
if (RTSP_RTP_PORT_MIN != 0) {
while (j <= RTSP_RTP_PORT_MAX) {
- snprintf(buf, sizeof(buf), "rtp://%s?localport=%d",
- host, j);
+ ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
+ "?localport=%d", j);
/* we will use two ports per rtp stream (rtp and rtcp) */
j += 2;
if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
@@ -1088,8 +1141,10 @@
snprintf(transport, sizeof(transport) - 1,
"%s/UDP;multicast", trans_pref);
}
- if (rt->server_type == RTSP_SERVER_REAL ||
- rt->server_type == RTSP_SERVER_WMS)
+ if (s->oformat) {
+ av_strlcat(transport, ";mode=receive", sizeof(transport));
+ } else if (rt->server_type == RTSP_SERVER_REAL ||
+ rt->server_type == RTSP_SERVER_WMS)
av_strlcat(transport, ";mode=play", sizeof(transport));
snprintf(cmd, sizeof(cmd),
"SETUP %s RTSP/1.0\r\n"
@@ -1104,7 +1159,7 @@
"RealChallenge2: %s, sd=%s\r\n",
rt->session_id, real_res, real_csum);
}
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
err = 1;
goto fail;
@@ -1143,13 +1198,19 @@
char url[1024];
/* XXX: also use address if specified */
- snprintf(url, sizeof(url), "rtp://%s:%d",
- host, reply->transports[0].server_port_min);
+ ff_url_join(url, sizeof(url), "rtp", NULL, host,
+ reply->transports[0].server_port_min, NULL);
if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
}
+ /* Try to initialize the connection state in a
+ * potential NAT router by sending dummy packets.
+ * RTP/RTCP dummy packets are used for RDT, too.
+ */
+ if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
+ rtp_send_punch_packets(rtsp_st->rtp_handle);
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
@@ -1166,8 +1227,8 @@
port = rtsp_st->sdp_port;
ttl = rtsp_st->sdp_ttl;
}
- snprintf(url, sizeof(url), "rtp://%s:%d?ttl=%d",
- inet_ntoa(in), port, ttl);
+ ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
+ port, "?ttl=%d", ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
@@ -1218,31 +1279,119 @@
rt->control_uri,
(double)rt->seek_timestamp / AV_TIME_BASE);
}
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
}
- rt->state = RTSP_STATE_PLAYING;
+ rt->state = RTSP_STATE_STREAMING;
return 0;
}
-static int rtsp_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
+static int rtsp_setup_input_streams(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
+ RTSPMessageHeader reply1, *reply = &reply1;
+ char cmd[1024];
+ unsigned char *content = NULL;
+ int ret;
+
+ /* describe the stream */
+ snprintf(cmd, sizeof(cmd),
+ "DESCRIBE %s RTSP/1.0\r\n"
+ "Accept: application/sdp\r\n",
+ s->filename);
+ if (rt->server_type == RTSP_SERVER_REAL) {
+ /**
+ * The Require: attribute is needed for proper streaming from
+ * Realmedia servers.
+ */
+ av_strlcat(cmd,
+ "Require: com.real.retain-entity-for-setup\r\n",
+ sizeof(cmd));
+ }
+ ff_rtsp_send_cmd(s, cmd, reply, &content);
+ if (!content)
+ return AVERROR_INVALIDDATA;
+ if (reply->status_code != RTSP_STATUS_OK) {
+ av_freep(&content);
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* now we got the SDP description, we parse it */
+ ret = sdp_parse(s, (const char *)content);
+ av_freep(&content);
+ if (ret < 0)
+ return AVERROR_INVALIDDATA;
+
+ return 0;
+}
+
+static int rtsp_setup_output_streams(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
+ RTSPMessageHeader reply1, *reply = &reply1;
+ char cmd[1024];
+ int i;
+ char *sdp;
+
+ /* Announce the stream */
+ snprintf(cmd, sizeof(cmd),
+ "ANNOUNCE %s RTSP/1.0\r\n"
+ "Content-Type: application/sdp\r\n",
+ s->filename);
+ sdp = av_mallocz(8192);
+ if (sdp == NULL)
+ return AVERROR(ENOMEM);
+ if (avf_sdp_create(&s, 1, sdp, 8192)) {
+ av_free(sdp);
+ return AVERROR_INVALIDDATA;
+ }
+ av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
+ ff_rtsp_send_cmd_with_content(s, cmd, reply, NULL, sdp, strlen(sdp));
+ av_free(sdp);
+ if (reply->status_code != RTSP_STATUS_OK)
+ return AVERROR_INVALIDDATA;
+
+ /* Set up the RTSPStreams for each AVStream */
+ for (i = 0; i < s->nb_streams; i++) {
+ RTSPStream *rtsp_st;
+ AVStream *st = s->streams[i];
+
+ rtsp_st = av_mallocz(sizeof(RTSPStream));
+ if (!rtsp_st)
+ return AVERROR(ENOMEM);
+ dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
+
+ st->priv_data = rtsp_st;
+ rtsp_st->stream_index = i;
+
+ av_strlcpy(rtsp_st->control_url, s->filename, sizeof(rtsp_st->control_url));
+ /* Note, this must match the relative uri set in the sdp content */
+ av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
+ "/streamid=%d", i);
+ }
+
+ return 0;
+}
+
+int ff_rtsp_connect(AVFormatContext *s)
+{
+ RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
char *option_list, *option, *filename;
URLContext *rtsp_hd;
- int port, ret, err;
+ int port, err;
RTSPMessageHeader reply1, *reply = &reply1;
- unsigned char *content = NULL;
int lower_transport_mask = 0;
char real_challenge[64];
+
+ if (!ff_network_init())
+ return AVERROR(EIO);
redirect:
/* extract hostname and port */
- url_split(NULL, 0, auth, sizeof(auth),
- host, sizeof(host), &port, path, sizeof(path), s->filename);
+ ff_url_split(NULL, 0, auth, sizeof(auth),
+ host, sizeof(host), &port, path, sizeof(path), s->filename);
if (*auth) {
int auth_len = strlen(auth), b64_len = ((auth_len + 2) / 3) * 4 + 1;
@@ -1286,8 +1435,19 @@
if (!lower_transport_mask)
lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
+ if (s->oformat) {
+ /* Only UDP output is supported at the moment. */
+ lower_transport_mask &= 1 << RTSP_LOWER_TRANSPORT_UDP;
+ if (!lower_transport_mask) {
+ av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
+ "only UDP is supported for output.\n");
+ err = AVERROR(EINVAL);
+ goto fail;
+ }
+ }
+
/* open the tcp connexion */
- snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
+ ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
err = AVERROR(EIO);
goto fail;
@@ -1318,7 +1478,7 @@
"CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
"GUID: 00000000-0000-0000-0000-000000000000\r\n",
sizeof(cmd));
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
err = AVERROR_INVALIDDATA;
goto fail;
@@ -1335,38 +1495,13 @@
break;
}
- /* describe the stream */
- snprintf(cmd, sizeof(cmd),
- "DESCRIBE %s RTSP/1.0\r\n"
- "Accept: application/sdp\r\n",
- s->filename);
- if (rt->server_type == RTSP_SERVER_REAL) {
- /**
- * The Require: attribute is needed for proper streaming from
- * Realmedia servers.
- */
- av_strlcat(cmd,
- "Require: com.real.retain-entity-for-setup\r\n",
- sizeof(cmd));
- }
- rtsp_send_cmd(s, cmd, reply, &content);
- if (!content) {
- err = AVERROR_INVALIDDATA;
+ if (s->iformat)
+ err = rtsp_setup_input_streams(s);
+ else
+ err = rtsp_setup_output_streams(s);
+ if (err)
goto fail;
- }
- if (reply->status_code != RTSP_STATUS_OK) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
- /* now we got the SDP description, we parse it */
- ret = sdp_parse(s, (const char *)content);
- av_freep(&content);
- if (ret < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
-
do {
int lower_transport = ff_log2_tab[lower_transport_mask &
~(lower_transport_mask - 1)];
@@ -1385,29 +1520,46 @@
rt->state = RTSP_STATE_IDLE;
rt->seek_timestamp = 0; /* default is to start stream at position zero */
- if (ap->initial_pause) {
- /* do not start immediately */
- } else {
- if (rtsp_read_play(s) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
- }
return 0;
fail:
- rtsp_close_streams(rt);
- av_freep(&content);
+ ff_rtsp_close_streams(s);
url_close(rt->rtsp_hd);
- if (reply->status_code >=300 && reply->status_code < 400) {
+ if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
av_strlcpy(s->filename, reply->location, sizeof(s->filename));
av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
reply->status_code,
s->filename);
goto redirect;
}
+ ff_network_close();
return err;
}
+#endif
+#if CONFIG_RTSP_DEMUXER
+static int rtsp_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ RTSPState *rt = s->priv_data;
+ int ret;
+
+ ret = ff_rtsp_connect(s);
+ if (ret)
+ return ret;
+
+ if (ap->initial_pause) {
+ /* do not start immediately */
+ } else {
+ if (rtsp_read_play(s) < 0) {
+ ff_rtsp_close_streams(s);
+ url_close(rt->rtsp_hd);
+ return AVERROR_INVALIDDATA;
+ }
+ }
+
+ return 0;
+}
+
static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size)
{
@@ -1460,9 +1612,9 @@
if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
RTSPMessageHeader reply;
- rtsp_read_reply(s, &reply, NULL, 0);
+ ff_rtsp_read_reply(s, &reply, NULL, 0);
/* XXX: parse message */
- if (rt->state != RTSP_STATE_PLAYING)
+ if (rt->state != RTSP_STATE_STREAMING)
return 0;
}
#endif
@@ -1484,13 +1636,13 @@
for (;;) {
RTSPMessageHeader reply;
- ret = rtsp_read_reply(s, &reply, NULL, 1);
+ ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
if (ret == -1)
return -1;
if (ret == 1) /* received '$' */
break;
/* XXX: parse message */
- if (rt->state != RTSP_STATE_PLAYING)
+ if (rt->state != RTSP_STATE_STREAMING)
return 0;
}
ret = url_read_complete(rt->rtsp_hd, buf, 3);
@@ -1600,7 +1752,7 @@
"SET_PARAMETER %s RTSP/1.0\r\n"
"Unsubscribe: %s\r\n",
rt->control_uri, rt->last_subscription);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 1;
@@ -1636,12 +1788,12 @@
}
}
av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK)
return AVERROR_INVALIDDATA;
rt->need_subscription = 0;
- if (rt->state == RTSP_STATE_PLAYING)
+ if (rt->state == RTSP_STATE_STREAMING)
rtsp_read_play (s);
}
}
@@ -1658,9 +1810,9 @@
snprintf(cmd, sizeof(cmd) - 1,
"GET_PARAMETER %s RTSP/1.0\r\n",
rt->control_uri);
- rtsp_send_cmd_async(s, cmd);
+ ff_rtsp_send_cmd_async(s, cmd);
} else {
- rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n");
+ ff_rtsp_send_cmd_async(s, "OPTIONS * RTSP/1.0\r\n");
}
}
@@ -1676,13 +1828,13 @@
rt = s->priv_data;
- if (rt->state != RTSP_STATE_PLAYING)
+ if (rt->state != RTSP_STATE_STREAMING)
return 0;
else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
snprintf(cmd, sizeof(cmd),
"PAUSE %s RTSP/1.0\r\n",
rt->control_uri);
- rtsp_send_cmd(s, cmd, reply, NULL);
+ ff_rtsp_send_cmd(s, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return -1;
}
@@ -1703,7 +1855,7 @@
default:
case RTSP_STATE_IDLE:
break;
- case RTSP_STATE_PLAYING:
+ case RTSP_STATE_STREAMING:
if (rtsp_read_pause(s) != 0)
return -1;
rt->state = RTSP_STATE_SEEKING;
@@ -1731,10 +1883,11 @@
snprintf(cmd, sizeof(cmd),
"TEARDOWN %s RTSP/1.0\r\n",
s->filename);
- rtsp_send_cmd_async(s, cmd);
+ ff_rtsp_send_cmd_async(s, cmd);
- rtsp_close_streams(rt);
+ ff_rtsp_close_streams(s);
url_close(rt->rtsp_hd);
+ ff_network_close();
return 0;
}
@@ -1782,6 +1935,9 @@
char *content;
char url[1024];
+ if (!ff_network_init())
+ return AVERROR(EIO);
+
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
@@ -1799,11 +1955,10 @@
for (i = 0; i < rt->nb_rtsp_streams; i++) {
rtsp_st = rt->rtsp_streams[i];
- snprintf(url, sizeof(url), "rtp://%s:%d?localport=%d&ttl=%d",
- inet_ntoa(rtsp_st->sdp_ip),
- rtsp_st->sdp_port,
- rtsp_st->sdp_port,
- rtsp_st->sdp_ttl);
+ ff_url_join(url, sizeof(url), "rtp", NULL,
+ inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
+ "?localport=%d&ttl=%d", rtsp_st->sdp_port,
+ rtsp_st->sdp_ttl);
if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
err = AVERROR_INVALIDDATA;
goto fail;
@@ -1813,14 +1968,15 @@
}
return 0;
fail:
- rtsp_close_streams(rt);
+ ff_rtsp_close_streams(s);
+ ff_network_close();
return err;
}
static int sdp_read_close(AVFormatContext *s)
{
- RTSPState *rt = s->priv_data;
- rtsp_close_streams(rt);
+ ff_rtsp_close_streams(s);
+ ff_network_close();
return 0;
}
« patched-ffmpeg-mt/libavcodec/mpeg4video_es_bsf.c ('K') | « patched-ffmpeg-mt/libavformat/rtsp.h ('k') | patched-ffmpeg-mt/libavformat/rtspenc.c » ('j') | no next file with comments »

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