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authorRomain Degez <romain.degez@smartjog.com>2005-05-26 07:47:51 +0000
committerMichael Niedermayer <michaelni@gmx.at>2005-05-26 07:47:51 +0000
commitd1ccf0e0a66bf9b09d86d799ca3b3616a14d2427 (patch)
treeb7163163fc4fff788efc969c1bce3811345c0037 /libavformat/rtsp.c
parent3072f0cb2ede227b803eee3b3e7d325991192858 (diff)
RTP/RTSP and MPEG4-AAC audio
- preliminary support for mpeg4-aac rtp payload (no interleaving support) - use udp transport as default (makes more sense with rtp, doesn't it ?) - some code factorization, so adding support for new rtp payload will be easier (I hope ;-) patch by (Romain DEGEZ: romain degez, smartjog com) Originally committed as revision 4306 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtsp.c')
-rw-r--r--libavformat/rtsp.c195
1 files changed, 141 insertions, 54 deletions
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 3261cf3d23..67c23aa4bc 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -66,22 +66,15 @@ typedef struct RTSPStream {
struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
int sdp_payload_type; /* payload type - only used in SDP */
+ rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
} RTSPStream;
static int rtsp_read_play(AVFormatContext *s);
/* XXX: currently, the only way to change the protocols consists in
changing this variable */
-#if 0
-int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_TCP) | (1 << RTSP_PROTOCOL_RTP_UDP) | (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST);
-#else
-/* try it if a proxy is used */
-int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_TCP);
-#endif
-/* if non zero, then set a range for RTP ports */
-int rtsp_rtp_port_min = 0;
-int rtsp_rtp_port_max = 0;
+int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
FFRTSPCallback *ff_rtsp_callback = NULL;
@@ -113,6 +106,8 @@ static void get_word_sep(char *buf, int buf_size, const char *sep,
char *q;
p = *pp;
+ if (*p == '/')
+ p++;
skip_spaces(&p);
q = buf;
while (!strchr(sep, *p) && *p != '\0') {
@@ -145,18 +140,67 @@ static void get_word(char *buf, int buf_size, const char **pp)
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
params>] */
-static int sdp_parse_rtpmap(AVCodecContext *codec, const char *p)
+static int sdp_parse_rtpmap(AVCodecContext *codec, int payload_type, const char *p)
{
char buf[256];
+ int i;
+ AVCodec *c;
+ char *c_name;
- /* codec name */
+ /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
+ see if we can handle this kind of payload */
get_word_sep(buf, sizeof(buf), "/", &p);
- if (!strcmp(buf, "MP4V-ES")) {
- codec->codec_id = CODEC_ID_MPEG4;
- return 0;
+ if (payload_type >= RTP_PT_PRIVATE) {
+ /* We are in dynmaic payload type case ... search into AVRtpDynamicPayloadTypes[] */
+ for (i = 0; AVRtpDynamicPayloadTypes[i].codec_id != CODEC_ID_NONE; ++i)
+ if (!strcmp(buf, AVRtpDynamicPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpDynamicPayloadTypes[i].codec_type)) {
+ codec->codec_id = AVRtpDynamicPayloadTypes[i].codec_id;
+ break;
+ }
} else {
- return -1;
+ /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
+ /* search into AVRtpPayloadTypes[] */
+ for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
+ if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
+ codec->codec_id = AVRtpPayloadTypes[i].codec_id;
+ break;
+ }
}
+
+ c = avcodec_find_decoder(codec->codec_id);
+ if (c && c->name)
+ c_name = (char *)c->name;
+ else
+ c_name = (char *)NULL;
+
+ if (c_name) {
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ switch (codec->codec_type) {
+ case CODEC_TYPE_AUDIO:
+ av_log(codec, 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;
+ }
+ av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
+ av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
+ break;
+ case CODEC_TYPE_VIDEO:
+ av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
+ break;
+ default:
+ break;
+ }
+ return 0;
+ }
+
+ return -1;
}
/* return the length and optionnaly the data */
@@ -188,11 +232,58 @@ static int hex_to_data(uint8_t *data, const char *p)
return len;
}
-static void sdp_parse_fmtp(AVCodecContext *codec, const char *p)
+static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
+{
+ switch (codec->codec_id) {
+ case CODEC_ID_MPEG4:
+ case CODEC_ID_MPEG4AAC:
+ if (!strcmp(attr, "config")) {
+ /* decode the hexa encoded parameter */
+ int len = hex_to_data(NULL, value);
+ codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!codec->extradata)
+ return;
+ codec->extradata_size = len;
+ hex_to_data(codec->extradata, value);
+ }
+ break;
+ default:
+ break;
+ }
+ return;
+}
+
+typedef struct attrname_map
+{
+ char *str;
+ uint16_t type;
+ uint32_t offset;
+} attrname_map_t;
+
+/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
+#define ATTR_NAME_TYPE_INT 0
+#define ATTR_NAME_TYPE_STR 1
+static attrname_map_t attr_names[]=
+{
+ {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
+ {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
+ {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
+ {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
+ {"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
+ {"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
+ {NULL, -1, -1},
+};
+
+/* parse a SDP line and save stream attributes */
+static void sdp_parse_fmtp(AVStream *st, const char *p)
{
char attr[256];
char value[4096];
- int len;
+ int i;
+
+ RTSPStream *rtsp_st = st->priv_data;
+ AVCodecContext *codec = &st->codec;
+ rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
/* loop on each attribute */
for(;;) {
@@ -205,25 +296,17 @@ static void sdp_parse_fmtp(AVCodecContext *codec, const char *p)
get_word_sep(value, sizeof(value), ";", &p);
if (*p == ';')
p++;
- /* handle MPEG4 video */
- switch(codec->codec_id) {
- case CODEC_ID_MPEG4:
- if (!strcmp(attr, "config")) {
- /* decode the hexa encoded parameter */
- len = hex_to_data(NULL, value);
- codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
- if (!codec->extradata)
- goto fail;
- codec->extradata_size = len;
- hex_to_data(codec->extradata, value);
- }
- break;
- default:
- /* ignore data for other codecs */
- break;
+ /* grab the codec extra_data from the config parameter of the fmtp line */
+ sdp_parse_fmtp_config(codec, attr, value);
+ /* Looking for a known attribute */
+ for (i = 0; attr_names[i].str; ++i) {
+ if (!strcasecmp(attr, attr_names[i].str)) {
+ if (attr_names[i].type == ATTR_NAME_TYPE_INT)
+ *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
+ else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
+ *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
+ }
}
- fail: ;
- // printf("'%s' = '%s'\n", attr, value);
}
}
@@ -314,7 +397,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
get_word(buf1, sizeof(buf1), &p); /* format list */
rtsp_st->sdp_payload_type = atoi(buf1);
- if (rtsp_st->sdp_payload_type == RTP_PT_MPEG2TS) {
+ if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
/* no corresponding stream */
} else {
st = av_new_stream(s, 0);
@@ -323,7 +406,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
st->priv_data = rtsp_st;
rtsp_st->stream_index = st->index;
st->codec.codec_type = codec_type;
- if (rtsp_st->sdp_payload_type < 96) {
+ if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
/* if standard payload type, we can find the codec right now */
rtp_get_codec_info(&st->codec, rtsp_st->sdp_payload_type);
}
@@ -355,7 +438,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
- sdp_parse_rtpmap(&st->codec, p);
+ sdp_parse_rtpmap(&st->codec, payload_type, p);
}
}
} else if (strstart(p, "fmtp:", &p)) {
@@ -366,7 +449,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
- sdp_parse_fmtp(&st->codec, p);
+ sdp_parse_fmtp(st, p);
}
}
}
@@ -715,7 +798,7 @@ static int rtsp_read_header(AVFormatContext *s,
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048];
URLContext *rtsp_hd;
- int port, i, ret, err;
+ int port, i, j, ret, err;
RTSPHeader reply1, *reply = &reply1;
unsigned char *content = NULL;
RTSPStream *rtsp_st;
@@ -763,7 +846,8 @@ static int rtsp_read_header(AVFormatContext *s,
/* for each stream, make the setup request */
/* XXX: we assume the same server is used for the control of each
RTSP stream */
- for(i=0;i<rt->nb_rtsp_streams;i++) {
+
+ for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
rtsp_st = rt->rtsp_streams[i];
@@ -774,22 +858,24 @@ static int rtsp_read_header(AVFormatContext *s,
/* RTP/UDP */
if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) {
char buf[256];
- int j;
/* first try in specified port range */
- if (rtsp_rtp_port_min != 0) {
- for(j=rtsp_rtp_port_min;j<=rtsp_rtp_port_max;j++) {
+ if (RTSP_RTP_PORT_MIN != 0) {
+ while(j <= RTSP_RTP_PORT_MAX) {
snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
- if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0)
+ if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0) {
+ j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
goto rtp_opened;
+ }
}
}
- /* then try on any port */
- if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
+/* then try on any port
+** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
+** err = AVERROR_INVALIDDATA;
+** goto fail;
+** }
+*/
rtp_opened:
port = rtp_get_local_port(rtsp_st->rtp_handle);
@@ -801,14 +887,14 @@ static int rtsp_read_header(AVFormatContext *s,
}
/* RTP/TCP */
- if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
+ else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
"RTP/AVP/TCP");
}
- if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
+ else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport),
@@ -887,7 +973,8 @@ static int rtsp_read_header(AVFormatContext *s,
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+ rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
+
if (!rtsp_st->rtp_ctx) {
err = AVERROR_NOMEM;
goto fail;
@@ -1233,7 +1320,7 @@ static int sdp_read_header(AVFormatContext *s,
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+ rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
if (!rtsp_st->rtp_ctx) {
err = AVERROR_NOMEM;
goto fail;