Implement RTMP output (publishing FLV stream to RTMP server).

Patch by Sergiy (piratfm at `do-no-evil-mail`.com)



git-svn-id: file:///var/local/repositories/ffmpeg/trunk@20731 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

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

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, \

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;
@@ -133,6 +145,75 @@ static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
 }
 
+/**
+ * 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.
@@ -147,13 +228,36 @@ 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);
+}
+
 /**
  * Generates 'play' call and sends it to the server, then pings the server
  * to start actual playing.
@@ -189,6 +293,30 @@ static void gen_play(URLContext *s, RTMPContext *rt)
     ff_rtmp_packet_destroy(&pkt);
 }
 
+/**
+ * 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.
  */
@@ -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)
 {
+    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 = {