* 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 "libavcodec/bytestream.h"
#include "libavutil/avstring.h"
+#include "libavutil/intfloat.h"
#include "libavutil/lfg.h"
+#include "libavutil/opt.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
+
+#define APP_MAX_LENGTH 128
+#define PLAYPATH_MAX_LENGTH 256
+#define TCURL_MAX_LENGTH 512
+#define FLASHVER_MAX_LENGTH 64
/** 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 */
typedef struct RTMPContext {
+ const AVClass *class;
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 *playpath; ///< stream identifier to play (with possible "mp4:" prefix)
+ int live; ///< 0: recorded, -1: live, -2: both
+ char *app; ///< name of 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
+ 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
+ int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
+ uint8_t flv_header[11]; ///< partial incoming flv packet header
+ int flv_header_bytes; ///< number of initialized bytes in flv_header
+ int nb_invokes; ///< keeps track of invoke messages
+ int create_stream_invoke; ///< invoke id for the create stream command
+ char* tcurl; ///< url of the target stream
+ char* flashver; ///< version of the flash plugin
+ char* swfurl; ///< url of the swf player
} RTMPContext;
#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
};
/**
- * 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)
+static void gen_connect(URLContext *s, RTMPContext *rt)
{
RTMPPacket pkt;
- uint8_t ver[32], *p;
- char tcurl[512];
+ uint8_t *p;
- 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_amf_write_string(&p, "connect");
- ff_amf_write_number(&p, 1.0);
+ ff_amf_write_number(&p, ++rt->nb_invokes);
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) {
+ 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_string(&p, rt->flashver);
+
+ if (rt->swfurl) {
+ ff_amf_write_field_name(&p, "swfUrl");
+ ff_amf_write_string(&p, rt->swfurl);
+ }
+
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);
+ ff_amf_write_string(&p, rt->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");
+ ff_amf_write_number(&p, 15.0);
+
+ /* Tell the server we support all the audio codecs except
+ * SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010)
+ * which are unused in the RTMP protocol implementation. */
+ ff_amf_write_field_name(&p, "audioCodecs");
+ ff_amf_write_number(&p, 4071.0);
+ 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, ++rt->nb_invokes);
+ 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, ++rt->nb_invokes);
+ 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 '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, ++rt->nb_invokes);
+ 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 '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)
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->nb_invokes);
+ ff_amf_write_null(&p);
+ rt->create_stream_invoke = rt->nb_invokes;
+
+ ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+}
+
+
+/**
+ * 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, ++rt->nb_invokes);
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
+ * 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)
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));
+ 29 + strlen(rt->playpath));
pkt.extra = rt->main_channel_id;
p = pkt.data;
ff_amf_write_string(&p, "play");
- ff_amf_write_number(&p, 0.0);
+ ff_amf_write_number(&p, ++rt->nb_invokes);
ff_amf_write_null(&p);
ff_amf_write_string(&p, rt->playpath);
+ ff_amf_write_number(&p, rt->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.
+ * 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, ++rt->nb_invokes);
+ 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)
{
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 server bandwidth message and send it to the server.
+ */
+static void gen_server_bw(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+
+ ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_SERVER_BW, 0, 4);
+ p = pkt.data;
+ bytestream_put_be32(&p, 2500000);
+ ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+}
+
+/**
+ * Generate check bandwidth message and send it to the server.
+ */
+static void gen_check_bw(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+
+ ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 21);
+
+ p = pkt.data;
+ ff_amf_write_string(&p, "_checkbw");
+ ff_amf_write_number(&p, ++rt->nb_invokes);
+ ff_amf_write_null(&p);
+
+ 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);
}
#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)
}
/**
- * 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)
}
/**
- * 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
}
/**
- * 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
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
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
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)) {
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 {
+ gen_server_bw(s, rt);
+ 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 = av_int2double(AV_RB64(pkt->data + 11));
+ if (pkt_id == rt->create_stream_invoke)
+ 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));
+ rt->main_channel_id = av_int2double(AV_RB64(pkt->data + 21));
+ }
+ if (rt->is_input) {
+ gen_play(s, rt);
+ } else {
+ gen_publish(s, rt);
}
- gen_play(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);
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;
+ } else if (!memcmp(pkt->data, "\002\000\010onBWDone", 11)) {
+ gen_check_bw(s, rt);
}
break;
}
}
/**
- * 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
{
RTMPContext *rt = s->priv_data;
int ret;
+ uint8_t *p;
+ const uint8_t *next;
+ uint32_t data_size;
+ uint32_t ts, cts, pts=0;
- for(;;) {
- RTMPPacket rpkt;
+ if (rt->state == STATE_STOPPED)
+ return AVERROR_EOF;
+
+ for (;;) {
+ RTMPPacket rpkt = { 0 };
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 && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
- uint8_t *p;
- uint32_t ts = rpkt.timestamp;
+ ts = rpkt.timestamp;
// generate packet header and put data into buffer for FLV demuxer
rt->flv_off = 0;
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;
}
ff_rtmp_packet_destroy(&rpkt);
}
- return 0;
}
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);
- av_free(rt);
+ ffurl_close(rt->stream);
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
*/
static int rtmp_open(URLContext *s, const char *uri, int flags)
{
- RTMPContext *rt;
- char proto[8], hostname[256], path[1024], app[128], *fname;
+ RTMPContext *rt = s->priv_data;
+ char proto[8], hostname[256], path[1024], *fname;
+ char *old_app;
uint8_t buf[2048];
int port;
int ret;
- rt = av_mallocz(sizeof(RTMPContext));
- if (!rt)
- return AVERROR(ENOMEM);
- s->priv_data = rt;
- rt->is_input = !(flags & URL_WRONLY);
+ 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) {
- av_log(LOG_CONTEXT, AV_LOG_ERROR, "Cannot open connection %s\n", buf);
+ if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
+ &s->interrupt_callback, NULL) < 0) {
+ av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
goto fail;
}
- if (!rt->is_input) {
- av_log(LOG_CONTEXT, AV_LOG_ERROR, "RTMP output is not supported yet.\n");
+ rt->state = STATE_START;
+ if (rtmp_handshake(s, rt))
goto fail;
- } else {
- 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;
+
+ // Keep the application name when it has been defined by the user.
+ old_app = rt->app;
+
+ rt->app = av_malloc(APP_MAX_LENGTH);
+ if (!rt->app) {
+ rtmp_close(s);
+ return AVERROR(ENOMEM);
+ }
+
+ //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 (old_app) {
+ // The name of application has been defined by the user, override it.
+ av_free(rt->app);
+ rt->app = old_app;
+ }
+
+ if (!rt->playpath) {
+ rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
+ if (!rt->playpath) {
+ rtmp_close(s);
+ return AVERROR(ENOMEM);
+ }
+
if (!strchr(fname, ':') &&
(!strcmp(fname + strlen(fname) - 4, ".f4v") ||
!strcmp(fname + strlen(fname) - 4, ".mp4"))) {
} else {
rt->playpath[0] = 0;
}
- strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
+ strncat(rt->playpath, fname, PLAYPATH_MAX_LENGTH - 5);
+ }
+
+ if (!rt->tcurl) {
+ rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
+ ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
+ port, "/%s", rt->app);
+ }
+
+ if (!rt->flashver) {
+ rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
+ if (rt->is_input) {
+ snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
+ RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
+ RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
+ } else {
+ snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
+ "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
+ }
+ }
- 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);
+ rt->client_report_size = 1048576;
+ rt->bytes_read = 0;
+ rt->last_bytes_read = 0;
- do {
- ret = get_packet(s, 1);
- } while (ret == EAGAIN);
- if (ret < 0)
- goto fail;
+ 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);
+
+ 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;
+ rt->skip_bytes = 13;
}
- 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;
buf += data_left;
size -= data_left;
rt->flv_off = rt->flv_size;
+ return data_left;
}
if ((ret = get_packet(s, 0)) < 0)
return ret;
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;
+
+ do {
+ if (rt->skip_bytes) {
+ int skip = FFMIN(rt->skip_bytes, size_temp);
+ buf_temp += skip;
+ size_temp -= skip;
+ rt->skip_bytes -= skip;
+ continue;
+ }
+
+ if (rt->flv_header_bytes < 11) {
+ const uint8_t *header = rt->flv_header;
+ int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
+ bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
+ rt->flv_header_bytes += copy;
+ size_temp -= copy;
+ if (rt->flv_header_bytes < 11)
+ break;
+
+ pkttype = bytestream_get_byte(&header);
+ pktsize = bytestream_get_be24(&header);
+ ts = bytestream_get_be24(&header);
+ ts |= bytestream_get_byte(&header) << 24;
+ bytestream_get_be24(&header);
+ 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;
+ size_temp = 0;
+ } else {
+ bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
+ size_temp -= rt->flv_size - rt->flv_off;
+ rt->flv_off += rt->flv_size - rt->flv_off;
+ }
+
+ if (rt->flv_off == rt->flv_size) {
+ rt->skip_bytes = 4;
+
+ 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;
+ rt->flv_header_bytes = 0;
+ }
+ } while (buf_temp - buf < size);
+ return size;
}
-URLProtocol rtmp_protocol = {
- "rtmp",
- rtmp_open,
- rtmp_read,
- rtmp_write,
- NULL, /* seek */
- rtmp_close,
+#define OFFSET(x) offsetof(RTMPContext, x)
+#define DEC AV_OPT_FLAG_DECODING_PARAM
+#define ENC AV_OPT_FLAG_ENCODING_PARAM
+
+static const AVOption rtmp_options[] = {
+ {"rtmp_app", "Name of application to connect to on the RTMP server", OFFSET(app), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+ {"rtmp_flashver", "Version of the Flash plugin used to run the SWF player.", OFFSET(flashver), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+ {"rtmp_live", "Specify that the media is a live stream.", OFFSET(live), AV_OPT_TYPE_INT, {-2}, INT_MIN, INT_MAX, DEC, "rtmp_live"},
+ {"any", "both", 0, AV_OPT_TYPE_CONST, {-2}, 0, 0, DEC, "rtmp_live"},
+ {"live", "live stream", 0, AV_OPT_TYPE_CONST, {-1}, 0, 0, DEC, "rtmp_live"},
+ {"recorded", "recorded stream", 0, AV_OPT_TYPE_CONST, {0}, 0, 0, DEC, "rtmp_live"},
+ {"rtmp_playpath", "Stream identifier to play or to publish", OFFSET(playpath), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+ {"rtmp_swfurl", "URL of the SWF player. By default no value will be sent", OFFSET(swfurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+ {"rtmp_tcurl", "URL of the target stream. Defaults to rtmp://host[:port]/app.", OFFSET(tcurl), AV_OPT_TYPE_STRING, {.str = NULL }, 0, 0, DEC|ENC},
+ { NULL },
+};
+
+static const AVClass rtmp_class = {
+ .class_name = "rtmp",
+ .item_name = av_default_item_name,
+ .option = rtmp_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+URLProtocol ff_rtmp_protocol = {
+ .name = "rtmp",
+ .url_open = rtmp_open,
+ .url_read = rtmp_read,
+ .url_write = rtmp_write,
+ .url_close = rtmp_close,
+ .priv_data_size = sizeof(RTMPContext),
+ .flags = URL_PROTOCOL_FLAG_NETWORK,
+ .priv_data_class= &rtmp_class,
};
projects / ffmpeg / blobdiff