udp: Fix receiving RTP data over multicast
[ffmpeg.git] / libavformat / rtmpproto.c
index 6d52214..70e4b14 100644 (file)
@@ -2,25 +2,25 @@
  * RTMP network protocol
  * Copyright (c) 2009 Kostya Shishkov
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
- * @file libavformat/rtmpproto.c
+ * @file
  * RTMP protocol
  */
 
 #include "libavutil/lfg.h"
 #include "libavutil/sha.h"
 #include "avformat.h"
+#include "internal.h"
 
 #include "network.h"
 
 #include "flv.h"
 #include "rtmp.h"
 #include "rtmppkt.h"
+#include "url.h"
 
-/* we can't use av_log() with URLContext yet... */
-#if LIBAVFORMAT_VERSION_MAJOR < 53
-#define LOG_CONTEXT NULL
-#else
-#define LOG_CONTEXT s
-#endif
+//#define DEBUG
 
 /** RTMP protocol handler state */
 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)
+    STATE_STOPPED,    ///< the broadcast has been stopped
 } ClientState;
 
 /** protocol handler context */
@@ -57,14 +58,18 @@ typedef struct RTMPContext {
     URLContext*   stream;                     ///< TCP stream used in interactions with RTMP server
     RTMPPacket    prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
     int           chunk_size;                 ///< size of the chunks RTMP packets are divided into
+    int           is_input;                   ///< input/output flag
     char          playpath[256];              ///< path to filename to play (with possible "mp4:" prefix)
+    char          app[128];                   ///< application
     ClientState   state;                      ///< current state
     int           main_channel_id;            ///< an additional channel ID which is used for some invocations
     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
-    uint32_t      video_ts;                   ///< current video timestamp in milliseconds
-    uint32_t      audio_ts;                   ///< current audio timestamp in milliseconds
+    RTMPPacket    out_pkt;                    ///< rtmp packet, created from flv a/v or metadata (for output)
+    uint32_t      client_report_size;         ///< number of bytes after which client should report to server
+    uint32_t      bytes_read;                 ///< number of bytes read from server
+    uint32_t      last_bytes_read;            ///< number of bytes read last reported to server
 } RTMPContext;
 
 #define PLAYER_KEY_OPEN_PART_LEN 30   ///< length of partial key used for first client digest signing
@@ -91,50 +96,128 @@ static const uint8_t rtmp_server_key[] = {
 };
 
 /**
- * Generates 'connect' call and sends it to the server.
+ * Generate 'connect' call and send it to the server.
  */
 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
-                        const char *host, int port, const char *app)
+                        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_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
+    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
     p = pkt.data;
 
-    snprintf(tcurl, sizeof(tcurl), "%s://%s:%d/%s", proto, host, port, app);
+    ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
     ff_amf_write_string(&p, "connect");
     ff_amf_write_number(&p, 1.0);
     ff_amf_write_object_start(&p);
     ff_amf_write_field_name(&p, "app");
-    ff_amf_write_string(&p, app);
+    ff_amf_write_string(&p, rt->app);
 
-    snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
-             RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
+    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);
-    ff_amf_write_field_name(&p, "fpad");
-    ff_amf_write_bool(&p, 0);
-    ff_amf_write_field_name(&p, "capabilities");
-    ff_amf_write_number(&p, 15.0);
-    ff_amf_write_field_name(&p, "audioCodecs");
-    ff_amf_write_number(&p, 1639.0);
-    ff_amf_write_field_name(&p, "videoCodecs");
-    ff_amf_write_number(&p, 252.0);
-    ff_amf_write_field_name(&p, "videoFunction");
-    ff_amf_write_number(&p, 1.0);
+    if (rt->is_input) {
+        ff_amf_write_field_name(&p, "fpad");
+        ff_amf_write_bool(&p, 0);
+        ff_amf_write_field_name(&p, "capabilities");
+        ff_amf_write_number(&p, 15.0);
+        ff_amf_write_field_name(&p, "audioCodecs");
+        ff_amf_write_number(&p, 1639.0);
+        ff_amf_write_field_name(&p, "videoCodecs");
+        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;
 
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+    ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
+ * Generate 'releaseStream' call and send 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(s, 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);
+}
+
+/**
+ * Generate 'FCPublish' call and send 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(s, 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 'createStream' call and sends it to the server. It should make
+ * Generate 'FCUnpublish' call and send 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(s, 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);
+}
+
+/**
+ * Generate 'createStream' call and send it to the server. It should make
  * the server allocate some channel for media streams.
  */
 static void gen_create_stream(URLContext *s, RTMPContext *rt)
@@ -142,20 +225,43 @@ static void gen_create_stream(URLContext *s, RTMPContext *rt)
     RTMPPacket pkt;
     uint8_t *p;
 
-    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Creating stream...\n");
-    ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0, 25);
+    av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
+    ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
 
     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 'play' call and sends it to the server, then pings the server
+ * Generate 'deleteStream' call and send 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(s, 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);
+}
+
+/**
+ * Generate 'play' call and send it to the server, then ping the server
  * to start actual playing.
  */
 static void gen_play(URLContext *s, RTMPContext *rt)
@@ -163,7 +269,7 @@ static void gen_play(URLContext *s, RTMPContext *rt)
     RTMPPacket pkt;
     uint8_t *p;
 
-    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
+    av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
     ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
                           20 + strlen(rt->playpath));
     pkt.extra = rt->main_channel_id;
@@ -190,7 +296,31 @@ static void gen_play(URLContext *s, RTMPContext *rt)
 }
 
 /**
- * Generates ping reply and sends it to the server.
+ * Generate 'publish' call and send it to the server.
+ */
+static void gen_publish(URLContext *s, RTMPContext *rt)
+{
+    RTMPPacket pkt;
+    uint8_t *p;
+
+    av_log(s, 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);
+}
+
+/**
+ * Generate ping reply and send it to the server.
  */
 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
 {
@@ -200,7 +330,22 @@ static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
     ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
     p = pkt.data;
     bytestream_put_be16(&p, 7);
-    bytestream_put_be32(&p, AV_RB32(ppkt->data+2) + 1);
+    bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
+    ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+    ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
+ * Generate report on bytes read so far and send it to the server.
+ */
+static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
+{
+    RTMPPacket pkt;
+    uint8_t *p;
+
+    ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
+    p = pkt.data;
+    bytestream_put_be32(&p, rt->bytes_read);
     ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
     ff_rtmp_packet_destroy(&pkt);
 }
@@ -210,7 +355,7 @@ static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
 #define HMAC_OPAD_VAL 0x5C
 
 /**
- * Calculates HMAC-SHA2 digest for RTMP handshake packets.
+ * Calculate HMAC-SHA2 digest for RTMP handshake packets.
  *
  * @param src    input buffer
  * @param len    input buffer length (should be 1536)
@@ -259,7 +404,7 @@ static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
 }
 
 /**
- * Puts HMAC-SHA2 digest of packet data (except for the bytes where this digest
+ * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
  * will be stored) into that packet.
  *
  * @param buf handshake data (1536 bytes)
@@ -280,7 +425,7 @@ static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
 }
 
 /**
- * Verifies that the received server response has the expected digest value.
+ * Verify that the received server response has the expected digest value.
  *
  * @param buf handshake data received from the server (1536 bytes)
  * @param off position to search digest offset from
@@ -304,7 +449,7 @@ static int rtmp_validate_digest(uint8_t *buf, int off)
 }
 
 /**
- * Performs handshake with the server by means of exchanging pseudorandom data
+ * Perform handshake with the server by means of exchanging pseudorandom data
  * signed with HMAC-SHA2 digest.
  *
  * @return 0 if handshake succeeds, negative value otherwise
@@ -326,7 +471,7 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
     int server_pos, client_pos;
     uint8_t digest[32];
 
-    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Handshaking...\n");
+    av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
 
     av_lfg_init(&rnd, 0xDEADC0DE);
     // generate handshake packet - 1536 bytes of pseudorandom data
@@ -334,57 +479,62 @@ static int rtmp_handshake(URLContext *s, RTMPContext *rt)
         tosend[i] = av_lfg_get(&rnd) >> 24;
     client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
 
-    url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
-    i = url_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
+    ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
+    i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
     if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
-        av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
+        av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
         return -1;
     }
-    i = url_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
+    i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
     if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
-        av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
+        av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
         return -1;
     }
 
-    av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
+    av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
            serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
 
-    server_pos = rtmp_validate_digest(serverdata + 1, 772);
-    if (!server_pos) {
-        server_pos = rtmp_validate_digest(serverdata + 1, 8);
+    if (rt->is_input && serverdata[5] >= 3) {
+        server_pos = rtmp_validate_digest(serverdata + 1, 772);
         if (!server_pos) {
-            av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server response validating failed\n");
+            server_pos = rtmp_validate_digest(serverdata + 1, 8);
+            if (!server_pos) {
+                av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
+                return -1;
+            }
+        }
+
+        rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
+                         rtmp_server_key, sizeof(rtmp_server_key),
+                         digest);
+        rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
+                         digest, 32,
+                         digest);
+        if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
+            av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
             return -1;
         }
-    }
 
-    rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
-                     rtmp_server_key, sizeof(rtmp_server_key),
-                     digest);
-    rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
-                     digest, 32,
-                     digest);
-    if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
-        av_log(LOG_CONTEXT, AV_LOG_ERROR, "Signature mismatch\n");
-        return -1;
+        for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
+            tosend[i] = av_lfg_get(&rnd) >> 24;
+        rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
+                         rtmp_player_key, sizeof(rtmp_player_key),
+                         digest);
+        rtmp_calc_digest(tosend,  RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
+                         digest, 32,
+                         tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
+
+        // write reply back to the server
+        ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
+    } else {
+        ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
     }
 
-    for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
-        tosend[i] = av_lfg_get(&rnd) >> 24;
-    rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
-                     rtmp_player_key, sizeof(rtmp_player_key),
-                     digest);
-    rtmp_calc_digest(tosend,  RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
-                     digest, 32,
-                     tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
-
-    // write reply back to the server
-    url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
     return 0;
 }
 
 /**
- * Parses received packet and may perform some action depending on
+ * Parse received packet and possibly perform some action depending on
  * the packet contents.
  * @return 0 for no errors, negative values for serious errors which prevent
  *         further communications, positive values for uncritical errors
@@ -394,25 +544,41 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
     int i, t;
     const uint8_t *data_end = pkt->data + pkt->data_size;
 
+#ifdef DEBUG
+    ff_rtmp_packet_dump(s, pkt);
+#endif
+
     switch (pkt->type) {
     case RTMP_PT_CHUNK_SIZE:
         if (pkt->data_size != 4) {
-            av_log(LOG_CONTEXT, AV_LOG_ERROR,
+            av_log(s, AV_LOG_ERROR,
                    "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);
+            av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
             return -1;
         }
-        av_log(LOG_CONTEXT, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
+        av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
         break;
     case RTMP_PT_PING:
         t = AV_RB16(pkt->data);
         if (t == 6)
             gen_pong(s, rt, pkt);
         break;
+    case RTMP_PT_CLIENT_BW:
+        if (pkt->data_size < 4) {
+            av_log(s, AV_LOG_ERROR,
+                   "Client bandwidth report packet is less than 4 bytes long (%d)\n",
+                   pkt->data_size);
+            return -1;
+        }
+        av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
+        rt->client_report_size = AV_RB32(pkt->data) >> 1;
+        break;
     case RTMP_PT_INVOKE:
         //TODO: check for the messages sent for wrong state?
         if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
@@ -420,29 +586,52 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
 
             if (!ff_amf_get_field_value(pkt->data + 9, data_end,
                                         "description", tmpstr, sizeof(tmpstr)))
-                av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
+                av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
             return -1;
         } 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]) {
-                    av_log(LOG_CONTEXT, AV_LOG_WARNING, "Unexpected reply on connect()\n");
+                    av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
                 } else {
                     rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
                 }
-                gen_play(s, rt);
+                if (rt->is_input) {
+                    gen_play(s, rt);
+                } else {
+                    gen_publish(s, rt);
+                }
                 rt->state = STATE_READY;
                 break;
             }
         } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
             const uint8_t* ptr = pkt->data + 11;
             uint8_t tmpstr[256];
-            int t;
 
             for (i = 0; i < 2; i++) {
                 t = ff_amf_tag_size(ptr, data_end);
@@ -455,15 +644,15 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
             if (!t && !strcmp(tmpstr, "error")) {
                 if (!ff_amf_get_field_value(ptr, data_end,
                                             "description", tmpstr, sizeof(tmpstr)))
-                    av_log(LOG_CONTEXT, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
+                    av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
                 return -1;
             }
             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.Play.Stop")) rt->state = STATE_STOPPED;
+            if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
+            if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
         }
         break;
     }
@@ -471,7 +660,7 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
 }
 
 /**
- * Interacts with the server by receiving and sending RTMP packets until
+ * Interact with the server by receiving and sending RTMP packets until
  * there is some significant data (media data or expected status notification).
  *
  * @param s          reading context
@@ -485,43 +674,52 @@ static int get_packet(URLContext *s, int for_header)
 {
     RTMPContext *rt = s->priv_data;
     int ret;
+    uint8_t *p;
+    const uint8_t *next;
+    uint32_t data_size;
+    uint32_t ts, cts, pts=0;
+
+    if (rt->state == STATE_STOPPED)
+        return AVERROR_EOF;
 
-    for(;;) {
+    for (;;) {
         RTMPPacket rpkt;
         if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
-                                       rt->chunk_size, rt->prev_pkt[0])) != 0) {
-            if (ret > 0) {
+                                       rt->chunk_size, rt->prev_pkt[0])) <= 0) {
+            if (ret == 0) {
                 return AVERROR(EAGAIN);
             } else {
                 return AVERROR(EIO);
             }
         }
+        rt->bytes_read += ret;
+        if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
+            av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
+            gen_bytes_read(s, rt, rpkt.timestamp + 1);
+            rt->last_bytes_read = rt->bytes_read;
+        }
 
         ret = rtmp_parse_result(s, rt, &rpkt);
         if (ret < 0) {//serious error in current packet
             ff_rtmp_packet_destroy(&rpkt);
             return -1;
         }
-        if (for_header && rt->state == STATE_PLAYING) {
+        if (rt->state == STATE_STOPPED) {
+            ff_rtmp_packet_destroy(&rpkt);
+            return AVERROR_EOF;
+        }
+        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;
         }
         if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
-            rpkt.type == RTMP_PT_NOTIFY) {
-            uint8_t *p;
-            uint32_t ts = rpkt.timestamp;
-
-            if (rpkt.type == RTMP_PT_VIDEO) {
-                rt->video_ts += rpkt.timestamp;
-                ts = rt->video_ts;
-            } else if (rpkt.type == RTMP_PT_AUDIO) {
-                rt->audio_ts += rpkt.timestamp;
-                ts = rt->audio_ts;
-            }
+           (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
+            ts = rpkt.timestamp;
+
             // generate packet header and put data into buffer for FLV demuxer
             rt->flv_off  = 0;
             rt->flv_size = rpkt.data_size + 15;
@@ -540,6 +738,23 @@ static int get_packet(URLContext *s, int for_header)
             rt->flv_off  = 0;
             rt->flv_size = rpkt.data_size;
             rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
+            /* rewrite timestamps */
+            next = rpkt.data;
+            ts = rpkt.timestamp;
+            while (next - rpkt.data < rpkt.data_size - 11) {
+                next++;
+                data_size = bytestream_get_be24(&next);
+                p=next;
+                cts = bytestream_get_be24(&next);
+                cts |= bytestream_get_byte(&next) << 24;
+                if (pts==0)
+                    pts=cts;
+                ts += cts - pts;
+                pts = cts;
+                bytestream_put_be24(&p, ts);
+                bytestream_put_byte(&p, ts >> 24);
+                next += data_size + 3 + 4;
+            }
             memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
             ff_rtmp_packet_destroy(&rpkt);
             return 0;
@@ -553,14 +768,24 @@ 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);
+        if (rt->state > STATE_FCPUBLISH)
+            gen_fcunpublish_stream(h, rt);
+    }
+    if (rt->state > STATE_HANDSHAKED)
+        gen_delete_stream(h, rt);
+
     av_freep(&rt->flv_data);
-    url_close(rt->stream);
+    ffurl_close(rt->stream);
     av_free(rt);
     return 0;
 }
 
 /**
- * Opens RTMP connection and verifies that the stream can be played.
+ * Open RTMP connection and verify that the stream can be played.
  *
  * URL syntax: rtmp://server[:port][/app][/playpath]
  *             where 'app' is first one or two directories in the path
@@ -571,85 +796,92 @@ static int rtmp_close(URLContext *h)
 static int rtmp_open(URLContext *s, const char *uri, int flags)
 {
     RTMPContext *rt;
-    char proto[8], hostname[256], path[1024], app[128], *fname;
+    char proto[8], hostname[256], path[1024], *fname;
     uint8_t buf[2048];
-    int port, is_input;
+    int port;
     int ret;
 
-    is_input = !(flags & URL_WRONLY);
-
     rt = av_mallocz(sizeof(RTMPContext));
     if (!rt)
         return AVERROR(ENOMEM);
     s->priv_data = rt;
+    rt->is_input = !(flags & AVIO_FLAG_WRITE);
 
-    url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
-              path, sizeof(path), s->filename);
+    av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
+                 path, sizeof(path), s->filename);
 
     if (port < 0)
         port = RTMP_DEFAULT_PORT;
-    snprintf(buf, sizeof(buf), "tcp://%s:%d", hostname, port);
+    ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
 
-    if (url_open(&rt->stream, buf, URL_RDWR) < 0)
+    if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE) < 0) {
+        av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
         goto fail;
+    }
 
-    if (!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;
+    rt->state = STATE_START;
+    if (rtmp_handshake(s, rt))
+        return -1;
 
-        rt->chunk_size = 128;
-        rt->state = STATE_HANDSHAKED;
-        //extract "app" part from path
-        if (!strncmp(path, "/ondemand/", 10)) {
-            fname = path + 10;
-            memcpy(app, "ondemand", 9);
+    rt->chunk_size = 128;
+    rt->state = STATE_HANDSHAKED;
+    //extract "app" part from path
+    if (!strncmp(path, "/ondemand/", 10)) {
+        fname = path + 10;
+        memcpy(rt->app, "ondemand", 9);
+    } else {
+        char *p = strchr(path + 1, '/');
+        if (!p) {
+            fname = path + 1;
+            rt->app[0] = '\0';
         } else {
-            char *p = strchr(path + 1, '/');
-            if (!p) {
-                fname = path + 1;
-                app[0] = '\0';
+            char *c = strchr(p + 1, ':');
+            fname = strchr(p + 1, '/');
+            if (!fname || c < fname) {
+                fname = p + 1;
+                av_strlcpy(rt->app, path + 1, p - path);
             } else {
-                char *c = strchr(p + 1, ':');
-                fname = strchr(p + 1, '/');
-                if (!fname || c < fname) {
-                    fname = p + 1;
-                    av_strlcpy(app, path + 1, p - path);
-                } else {
-                    fname++;
-                    av_strlcpy(app, path + 1, fname - path - 1);
-                }
+                fname++;
+                av_strlcpy(rt->app, path + 1, fname - path - 1);
             }
         }
-        if (!strchr(fname, ':') &&
-            (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
-             !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
-            memcpy(rt->playpath, "mp4:", 5);
-        } else {
-            rt->playpath[0] = 0;
-        }
-        strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
+    }
+    if (!strchr(fname, ':') &&
+        (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
+         !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
+        memcpy(rt->playpath, "mp4:", 5);
+    } else {
+        rt->playpath[0] = 0;
+    }
+    strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
+
+    rt->client_report_size = 1048576;
+    rt->bytes_read = 0;
+    rt->last_bytes_read = 0;
 
-        av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
-               proto, path, app, rt->playpath);
-        gen_connect(s, rt, proto, hostname, port, app);
+    av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
+           proto, path, rt->app, rt->playpath);
+    gen_connect(s, rt, proto, hostname, port);
 
-        do {
-            ret = get_packet(s, 1);
-        } while (ret == EAGAIN);
-        if (ret < 0)
-            goto fail;
+    do {
+        ret = get_packet(s, 1);
+    } 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);
+    s->max_packet_size = rt->stream->max_packet_size;
     s->is_streamed     = 1;
     return 0;
 
@@ -677,6 +909,7 @@ static int rtmp_read(URLContext *s, uint8_t *buf, int size)
             buf  += data_left;
             size -= data_left;
             rt->flv_off = rt->flv_size;
+            return data_left;
         }
         if ((ret = get_packet(s, 0)) < 0)
            return ret;
@@ -684,16 +917,76 @@ static int rtmp_read(URLContext *s, uint8_t *buf, int size)
     return orig_size;
 }
 
-static int rtmp_write(URLContext *h, uint8_t *buf, int size)
+static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
 {
-    return 0;
+    RTMPContext *rt = s->priv_data;
+    int size_temp = size;
+    int pktsize, pkttype;
+    uint32_t ts;
+    const uint8_t *buf_temp = buf;
+
+    if (size < 11) {
+        av_log(s, 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 = {
-    "rtmp",
-    rtmp_open,
-    rtmp_read,
-    rtmp_write,
-    NULL, /* seek */
-    rtmp_close,
+URLProtocol ff_rtmp_protocol = {
+    .name      = "rtmp",
+    .url_open  = rtmp_open,
+    .url_read  = rtmp_read,
+    .url_write = rtmp_write,
+    .url_close = rtmp_close,
 };