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authorSergiy <piratfm@gmail.com>2009-12-04 16:52:16 +0000
committerKostya Shishkov <kostya.shishkov@gmail.com>2009-12-04 16:52:16 +0000
commit6bf22e18d1357f11048902e2c5ac9f814cd123fa (patch)
tree8ab686f0da2b5f4afa99de6a60fa1ed1502e0acf
parentb83ccbffe9b109fcd18dbd178d6b4f300e6d6799 (diff)
downloadffmpeg-6bf22e18d1357f11048902e2c5ac9f814cd123fa.tar.gz
Implement RTMP output (publishing FLV stream to RTMP server).
Patch by Sergiy (piratfm at `do-no-evil-mail`.com) Originally committed as revision 20731 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--doc/general.texi3
-rw-r--r--libavformat/avformat.h2
-rw-r--r--libavformat/rtmpproto.c254
3 files changed, 244 insertions, 15 deletions
diff --git a/doc/general.texi b/doc/general.texi
index 09cd03a24a..17c080488b 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -201,7 +201,8 @@ library:
@item RL2 @tab @tab X
@tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie @tab @tab X
-@item RTMP @tab @tab X
+@item RTMP @tab X @tab X
+ @tab Output is performed by publishing stream to RTMP server
@item RTP @tab @tab X
@item RTSP @tab @tab X
@item SDP @tab @tab X
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index d0d9b8d221..507a491104 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -22,7 +22,7 @@
#define AVFORMAT_AVFORMAT_H
#define LIBAVFORMAT_VERSION_MAJOR 52
-#define LIBAVFORMAT_VERSION_MINOR 40
+#define LIBAVFORMAT_VERSION_MINOR 41
#define LIBAVFORMAT_VERSION_MICRO 0
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
diff --git a/libavformat/rtmpproto.c b/libavformat/rtmpproto.c
index e8f5c2df9f..837905cd68 100644
--- a/libavformat/rtmpproto.c
+++ b/libavformat/rtmpproto.c
@@ -47,9 +47,12 @@
typedef enum {
STATE_START, ///< client has not done anything yet
STATE_HANDSHAKED, ///< client has performed handshake
+ STATE_RELEASING, ///< client releasing stream before publish it (for output)
+ STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
STATE_CONNECTING, ///< client connected to server successfully
STATE_READY, ///< client has sent all needed commands and waits for server reply
STATE_PLAYING, ///< client has started receiving multimedia data from server
+ STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
} ClientState;
/** protocol handler context */
@@ -65,6 +68,7 @@ typedef struct RTMPContext {
uint8_t* flv_data; ///< buffer with data for demuxer
int flv_size; ///< current buffer size
int flv_off; ///< number of bytes read from current buffer
+ RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
} RTMPContext;
#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
@@ -97,7 +101,7 @@ static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
const char *host, int port)
{
RTMPPacket pkt;
- uint8_t ver[32], *p;
+ uint8_t ver[64], *p;
char tcurl[512];
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
@@ -110,12 +114,19 @@ static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
ff_amf_write_field_name(&p, "app");
ff_amf_write_string(&p, rt->app);
+ if (rt->is_input) {
snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
+ } else {
+ snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
+ ff_amf_write_field_name(&p, "type");
+ ff_amf_write_string(&p, "nonprivate");
+ }
ff_amf_write_field_name(&p, "flashVer");
ff_amf_write_string(&p, ver);
ff_amf_write_field_name(&p, "tcUrl");
ff_amf_write_string(&p, tcurl);
+ if (rt->is_input) {
ff_amf_write_field_name(&p, "fpad");
ff_amf_write_bool(&p, 0);
ff_amf_write_field_name(&p, "capabilities");
@@ -126,6 +137,7 @@ static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
ff_amf_write_number(&p, 252.0);
ff_amf_write_field_name(&p, "videoFunction");
ff_amf_write_number(&p, 1.0);
+ }
ff_amf_write_object_end(&p);
pkt.data_size = p - pkt.data;
@@ -134,6 +146,75 @@ static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
}
/**
+ * Generates 'releaseStream' call and sends it to the server. It should make
+ * the server release some channel for media streams.
+ */
+static void gen_release_stream(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+
+ ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
+ 29 + strlen(rt->playpath));
+
+ av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Releasing stream...\n");
+ p = pkt.data;
+ ff_amf_write_string(&p, "releaseStream");
+ ff_amf_write_number(&p, 2.0);
+ ff_amf_write_null(&p);
+ ff_amf_write_string(&p, rt->playpath);
+
+ ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
+ * Generates 'FCPublish' call and sends it to the server. It should make
+ * the server preapare for receiving media streams.
+ */
+static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+
+ ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
+ 25 + strlen(rt->playpath));
+
+ av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FCPublish stream...\n");
+ p = pkt.data;
+ ff_amf_write_string(&p, "FCPublish");
+ ff_amf_write_number(&p, 3.0);
+ ff_amf_write_null(&p);
+ ff_amf_write_string(&p, rt->playpath);
+
+ ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
+ * Generates 'FCUnpublish' call and sends it to the server. It should make
+ * the server destroy stream.
+ */
+static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+
+ ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
+ 27 + strlen(rt->playpath));
+
+ av_log(LOG_CONTEXT, AV_LOG_DEBUG, "UnPublishing stream...\n");
+ p = pkt.data;
+ ff_amf_write_string(&p, "FCUnpublish");
+ ff_amf_write_number(&p, 5.0);
+ ff_amf_write_null(&p);
+ ff_amf_write_string(&p, rt->playpath);
+
+ ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
* Generates 'createStream' call and sends it to the server. It should make
* the server allocate some channel for media streams.
*/
@@ -147,8 +228,31 @@ static void gen_create_stream(URLContext *s, RTMPContext *rt)
p = pkt.data;
ff_amf_write_string(&p, "createStream");
- ff_amf_write_number(&p, 3.0);
+ ff_amf_write_number(&p, rt->is_input ? 3.0 : 4.0);
+ ff_amf_write_null(&p);
+
+ ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+}
+
+
+/**
+ * Generates 'deleteStream' call and sends it to the server. It should make
+ * the server remove some channel for media streams.
+ */
+static void gen_delete_stream(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+
+ av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Deleting stream...\n");
+ ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
+
+ p = pkt.data;
+ ff_amf_write_string(&p, "deleteStream");
+ ff_amf_write_number(&p, 0.0);
ff_amf_write_null(&p);
+ ff_amf_write_number(&p, rt->main_channel_id);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
@@ -190,6 +294,30 @@ static void gen_play(URLContext *s, RTMPContext *rt)
}
/**
+ * Generates 'publish' call and sends it to the server.
+ */
+static void gen_publish(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+
+ av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
+ ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
+ 30 + strlen(rt->playpath));
+ pkt.extra = rt->main_channel_id;
+
+ p = pkt.data;
+ ff_amf_write_string(&p, "publish");
+ ff_amf_write_number(&p, 0.0);
+ ff_amf_write_null(&p);
+ ff_amf_write_string(&p, rt->playpath);
+ ff_amf_write_string(&p, "live");
+
+ ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
* Generates ping reply and sends it to the server.
*/
static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
@@ -349,6 +477,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
+ if (rt->is_input) {
server_pos = rtmp_validate_digest(serverdata + 1, 772);
if (!server_pos) {
server_pos = rtmp_validate_digest(serverdata + 1, 8);
@@ -380,6 +509,10 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
// write reply back to the server
url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
+ } else {
+ url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
+ }
+
return 0;
}
@@ -401,6 +534,8 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
"Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
return -1;
}
+ if (!rt->is_input)
+ ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
rt->chunk_size = AV_RB32(pkt->data);
if (rt->chunk_size <= 0) {
av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
@@ -425,9 +560,29 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
} else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
switch (rt->state) {
case STATE_HANDSHAKED:
+ if (!rt->is_input) {
+ gen_release_stream(s, rt);
+ gen_fcpublish_stream(s, rt);
+ rt->state = STATE_RELEASING;
+ } else {
+ rt->state = STATE_CONNECTING;
+ }
gen_create_stream(s, rt);
+ break;
+ case STATE_FCPUBLISH:
rt->state = STATE_CONNECTING;
break;
+ case STATE_RELEASING:
+ rt->state = STATE_FCPUBLISH;
+ /* hack for Wowza Media Server, it does not send result for
+ * releaseStream and FCPublish calls */
+ if (!pkt->data[10]) {
+ int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
+ if (pkt_id == 4)
+ rt->state = STATE_CONNECTING;
+ }
+ if(rt->state != STATE_CONNECTING)
+ break;
case STATE_CONNECTING:
//extract a number from the result
if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
@@ -435,7 +590,11 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
} else {
rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
}
+ if (rt->is_input) {
gen_play(s, rt);
+ } else {
+ gen_publish(s, rt);
+ }
rt->state = STATE_READY;
break;
}
@@ -459,10 +618,8 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
}
t = ff_amf_get_field_value(ptr, data_end,
"code", tmpstr, sizeof(tmpstr));
- if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) {
- rt->state = STATE_PLAYING;
- return 0;
- }
+ if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
+ if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
}
break;
}
@@ -501,11 +658,11 @@ static int get_packet(URLContext *s, int for_header)
ff_rtmp_packet_destroy(&rpkt);
return -1;
}
- if (for_header && rt->state == STATE_PLAYING) {
+ if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
ff_rtmp_packet_destroy(&rpkt);
return 0;
}
- if (!rpkt.data_size) {
+ if (!rpkt.data_size || !rt->is_input) {
ff_rtmp_packet_destroy(&rpkt);
continue;
}
@@ -545,6 +702,14 @@ static int rtmp_close(URLContext *h)
{
RTMPContext *rt = h->priv_data;
+ if(!rt->is_input) {
+ rt->flv_data = NULL;
+ if (rt->out_pkt.data_size)
+ ff_rtmp_packet_destroy(&rt->out_pkt);
+ gen_fcunpublish_stream(h, rt);
+ }
+ gen_delete_stream(h, rt);
+
av_freep(&rt->flv_data);
url_close(rt->stream);
av_free(rt);
@@ -586,10 +751,6 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
goto fail;
}
- if (!rt->is_input) {
- av_log(LOG_CONTEXT, AV_LOG_ERROR, "RTMP output is not supported yet.\n");
- goto fail;
- } else {
rt->state = STATE_START;
if (rtmp_handshake(s, rt))
return -1;
@@ -635,11 +796,17 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
} while (ret == EAGAIN);
if (ret < 0)
goto fail;
+
+ if (rt->is_input) {
// generate FLV header for demuxer
rt->flv_size = 13;
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
rt->flv_off = 0;
memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
+ } else {
+ rt->flv_size = 0;
+ rt->flv_data = NULL;
+ rt->flv_off = 0;
}
s->max_packet_size = url_get_max_packet_size(rt->stream);
@@ -679,7 +846,68 @@ static int rtmp_read(URLContext *s, uint8_t *buf, int size)
static int rtmp_write(URLContext *h, uint8_t *buf, int size)
{
- return 0;
+ RTMPContext *rt = h->priv_data;
+ int size_temp = size;
+ int pktsize, pkttype;
+ uint32_t ts;
+ const uint8_t *buf_temp = buf;
+
+ if (size < 11) {
+ av_log(LOG_CONTEXT, AV_LOG_DEBUG, "FLV packet too small %d\n", size);
+ return 0;
+ }
+
+ do {
+ if (!rt->flv_off) {
+ //skip flv header
+ if (buf_temp[0] == 'F' && buf_temp[1] == 'L' && buf_temp[2] == 'V') {
+ buf_temp += 9 + 4;
+ size_temp -= 9 + 4;
+ }
+
+ pkttype = bytestream_get_byte(&buf_temp);
+ pktsize = bytestream_get_be24(&buf_temp);
+ ts = bytestream_get_be24(&buf_temp);
+ ts |= bytestream_get_byte(&buf_temp) << 24;
+ bytestream_get_be24(&buf_temp);
+ size_temp -= 11;
+ rt->flv_size = pktsize;
+
+ //force 12bytes header
+ if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
+ pkttype == RTMP_PT_NOTIFY) {
+ if (pkttype == RTMP_PT_NOTIFY)
+ pktsize += 16;
+ rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
+ }
+
+ //this can be a big packet, it's better to send it right here
+ ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
+ rt->out_pkt.extra = rt->main_channel_id;
+ rt->flv_data = rt->out_pkt.data;
+
+ if (pkttype == RTMP_PT_NOTIFY)
+ ff_amf_write_string(&rt->flv_data, "@setDataFrame");
+ }
+
+ if (rt->flv_size - rt->flv_off > size_temp) {
+ bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
+ rt->flv_off += size_temp;
+ } else {
+ bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
+ rt->flv_off += rt->flv_size - rt->flv_off;
+ }
+
+ if (rt->flv_off == rt->flv_size) {
+ bytestream_get_be32(&buf_temp);
+
+ ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&rt->out_pkt);
+ rt->flv_size = 0;
+ rt->flv_off = 0;
+ }
+ } while (buf_temp - buf < size_temp);
+ return size;
}
URLProtocol rtmp_protocol = {