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

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

Originally committed as revision 20731 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Sergiy
2009-12-04 16:52:16 +00:00
committed by Kostya Shishkov
parent b83ccbffe9
commit 6bf22e18d1
3 changed files with 244 additions and 15 deletions
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3
doc/general.texi
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@@ -201,7 +201,8 @@ library:
@item RL2 @tab @tab X @item RL2 @tab @tab X
@tab Audio and video format used in some games by Entertainment Software Partners. @tab Audio and video format used in some games by Entertainment Software Partners.
@item RPL/ARMovie @tab @tab X @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 RTP @tab @tab X
@item RTSP @tab @tab X @item RTSP @tab @tab X
@item SDP @tab @tab X @item SDP @tab @tab X

2
libavformat/avformat.h
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@@ -22,7 +22,7 @@
#define AVFORMAT_AVFORMAT_H #define AVFORMAT_AVFORMAT_H
#define LIBAVFORMAT_VERSION_MAJOR 52 #define LIBAVFORMAT_VERSION_MAJOR 52
#define LIBAVFORMAT_VERSION_MINOR 40 #define LIBAVFORMAT_VERSION_MINOR 41
#define LIBAVFORMAT_VERSION_MICRO 0 #define LIBAVFORMAT_VERSION_MICRO 0
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \ #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \

254
libavformat/rtmpproto.c
View File

@@ -47,9 +47,12 @@
typedef enum { typedef enum {
STATE_START, ///< client has not done anything yet STATE_START, ///< client has not done anything yet
STATE_HANDSHAKED, ///< client has performed handshake 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_CONNECTING, ///< client connected to server successfully
STATE_READY, ///< client has sent all needed commands and waits for server reply STATE_READY, ///< client has sent all needed commands and waits for server reply
STATE_PLAYING, ///< client has started receiving multimedia data from server STATE_PLAYING, ///< client has started receiving multimedia data from server
STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
} ClientState; } ClientState;
/** protocol handler context */ /** protocol handler context */
@@ -65,6 +68,7 @@ typedef struct RTMPContext {
uint8_t* flv_data; ///< buffer with data for demuxer uint8_t* flv_data; ///< buffer with data for demuxer
int flv_size; ///< current buffer size int flv_size; ///< current buffer size
int flv_off; ///< number of bytes read from current buffer 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; } RTMPContext;
#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing #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) const char *host, int port)
{ {
RTMPPacket pkt; RTMPPacket pkt;
uint8_t ver[32], *p; uint8_t ver[64], *p;
char tcurl[512]; char tcurl[512];
ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096); 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_field_name(&p, "app");
ff_amf_write_string(&p, rt->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, snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4); 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_field_name(&p, "flashVer");
ff_amf_write_string(&p, ver); ff_amf_write_string(&p, ver);
ff_amf_write_field_name(&p, "tcUrl"); ff_amf_write_field_name(&p, "tcUrl");
ff_amf_write_string(&p, tcurl); ff_amf_write_string(&p, tcurl);
if (rt->is_input) {
ff_amf_write_field_name(&p, "fpad"); ff_amf_write_field_name(&p, "fpad");
ff_amf_write_bool(&p, 0); ff_amf_write_bool(&p, 0);
ff_amf_write_field_name(&p, "capabilities"); 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_number(&p, 252.0);
ff_amf_write_field_name(&p, "videoFunction"); ff_amf_write_field_name(&p, "videoFunction");
ff_amf_write_number(&p, 1.0); ff_amf_write_number(&p, 1.0);
}
ff_amf_write_object_end(&p); ff_amf_write_object_end(&p);
pkt.data_size = p - pkt.data; 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]); 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 * Generates 'createStream' call and sends it to the server. It should make
* the server allocate some channel for media streams. * the server allocate some channel for media streams.
@@ -147,13 +228,36 @@ static void gen_create_stream(URLContext *s, RTMPContext *rt)
p = pkt.data; p = pkt.data;
ff_amf_write_string(&p, "createStream"); 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_amf_write_null(&p);
ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]); ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt); 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 * Generates 'play' call and sends it to the server, then pings the server
* to start actual playing. * to start actual playing.
@@ -189,6 +293,30 @@ static void gen_play(URLContext *s, RTMPContext *rt)
ff_rtmp_packet_destroy(&pkt); 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. * 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", av_log(LOG_CONTEXT, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
serverdata[5], serverdata[6], serverdata[7], serverdata[8]); serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
if (rt->is_input) {
server_pos = rtmp_validate_digest(serverdata + 1, 772); server_pos = rtmp_validate_digest(serverdata + 1, 772);
if (!server_pos) { if (!server_pos) {
server_pos = rtmp_validate_digest(serverdata + 1, 8); 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 // write reply back to the server
url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE); url_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
} else {
url_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
}
return 0; 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); "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
return -1; 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); rt->chunk_size = AV_RB32(pkt->data);
if (rt->chunk_size <= 0) { if (rt->chunk_size <= 0) {
av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size); av_log(LOG_CONTEXT, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
@@ -425,8 +560,28 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
} else if (!memcmp(pkt->data, "\002\000\007_result", 10)) { } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
switch (rt->state) { switch (rt->state) {
case STATE_HANDSHAKED: case STATE_HANDSHAKED:
gen_create_stream(s, rt); if (!rt->is_input) {
gen_release_stream(s, rt);
gen_fcpublish_stream(s, rt);
rt->state = STATE_RELEASING;
} else {
rt->state = STATE_CONNECTING; 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; break;
case STATE_CONNECTING: case STATE_CONNECTING:
//extract a number from the result //extract a number from the result
@@ -435,7 +590,11 @@ static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
} else { } else {
rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21)); rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
} }
if (rt->is_input) {
gen_play(s, rt); gen_play(s, rt);
} else {
gen_publish(s, rt);
}
rt->state = STATE_READY; rt->state = STATE_READY;
break; 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, t = ff_amf_get_field_value(ptr, data_end,
"code", tmpstr, sizeof(tmpstr)); "code", tmpstr, sizeof(tmpstr));
if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) { if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
rt->state = STATE_PLAYING; if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
return 0;
}
} }
break; break;
} }
@@ -501,11 +658,11 @@ static int get_packet(URLContext *s, int for_header)
ff_rtmp_packet_destroy(&rpkt); ff_rtmp_packet_destroy(&rpkt);
return -1; 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); ff_rtmp_packet_destroy(&rpkt);
return 0; return 0;
} }
if (!rpkt.data_size) { if (!rpkt.data_size || !rt->is_input) {
ff_rtmp_packet_destroy(&rpkt); ff_rtmp_packet_destroy(&rpkt);
continue; continue;
} }
@@ -545,6 +702,14 @@ static int rtmp_close(URLContext *h)
{ {
RTMPContext *rt = h->priv_data; 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); av_freep(&rt->flv_data);
url_close(rt->stream); url_close(rt->stream);
av_free(rt); av_free(rt);
@@ -586,10 +751,6 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
goto fail; 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; rt->state = STATE_START;
if (rtmp_handshake(s, rt)) if (rtmp_handshake(s, rt))
return -1; return -1;
@@ -635,11 +796,17 @@ static int rtmp_open(URLContext *s, const char *uri, int flags)
} while (ret == EAGAIN); } while (ret == EAGAIN);
if (ret < 0) if (ret < 0)
goto fail; goto fail;
if (rt->is_input) {
// generate FLV header for demuxer // generate FLV header for demuxer
rt->flv_size = 13; rt->flv_size = 13;
rt->flv_data = av_realloc(rt->flv_data, rt->flv_size); rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
rt->flv_off = 0; rt->flv_off = 0;
memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size); 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 = 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) 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; 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 = { URLProtocol rtmp_protocol = {