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
+ int flv_nb_packets; ///< number of flv packets published
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
char* tcurl; ///< url of the target stream
char* flashver; ///< version of the flash plugin
char* swfurl; ///< url of the swf player
+ int server_bw; ///< server bandwidth
+ int client_buffer_time; ///< client buffer time in ms
+ int flush_interval; ///< number of packets flushed in the same request (RTMPT only)
} RTMPContext;
#define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
return ret;
}
+/**
+ * Generate client buffer time and send it to the server.
+ */
+static int gen_buffer_time(URLContext *s, RTMPContext *rt)
+{
+ RTMPPacket pkt;
+ uint8_t *p;
+ int ret;
+
+ if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
+ 1, 10)) < 0)
+ return ret;
+
+ p = pkt.data;
+ bytestream_put_be16(&p, 3);
+ bytestream_put_be32(&p, rt->main_channel_id);
+ bytestream_put_be32(&p, rt->client_buffer_time);
+
+ ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
+ rt->prev_pkt[1]);
+ ff_rtmp_packet_destroy(&pkt);
+
+ return ret;
+}
+
/**
* Generate 'play' call and send it to the server, then ping the server
* to start actual playing.
rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
- if (ret < 0)
- return ret;
-
- // set client buffer time disguised in ping packet
- if ((ret = ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING,
- 1, 10)) < 0)
- return ret;
-
- p = pkt.data;
- bytestream_put_be16(&p, 3);
- bytestream_put_be32(&p, 1);
- bytestream_put_be32(&p, 256); //TODO: what is a good value here?
-
- ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
- rt->prev_pkt[1]);
- ff_rtmp_packet_destroy(&pkt);
-
return ret;
}
return ret;
p = pkt.data;
- bytestream_put_be32(&p, 2500000);
+ bytestream_put_be32(&p, rt->server_bw);
ret = ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size,
rt->prev_pkt[1]);
ff_rtmp_packet_destroy(&pkt);
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_SERVER_BW:
+ rt->server_bw = AV_RB32(pkt->data);
+ if (rt->server_bw <= 0) {
+ av_log(s, AV_LOG_ERROR, "Incorrect server bandwidth %d\n", rt->server_bw);
+ return AVERROR(EINVAL);
+ }
+ av_log(s, AV_LOG_DEBUG, "Server bandwidth = %d\n", rt->server_bw);
+ break;
case RTMP_PT_INVOKE:
//TODO: check for the messages sent for wrong state?
if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
if (rt->is_input) {
if ((ret = gen_play(s, rt)) < 0)
return ret;
+ if ((ret = gen_buffer_time(s, rt)) < 0)
+ return ret;
} else {
if ((ret = gen_publish(s, rt)) < 0)
return ret;
return ret;
}
break;
+ default:
+ av_log(s, AV_LOG_VERBOSE, "Unknown packet type received 0x%02X\n", pkt->type);
+ break;
}
return 0;
}
av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
path, sizeof(path), s->filename);
- if (port < 0)
- port = RTMP_DEFAULT_PORT;
- ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
+ if (!strcmp(proto, "rtmpt")) {
+ /* open the http tunneling connection */
+ ff_url_join(buf, sizeof(buf), "rtmphttp", NULL, hostname, port, NULL);
+ } else {
+ /* open the tcp connection */
+ if (port < 0)
+ port = RTMP_DEFAULT_PORT;
+ ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
+ }
if ((ret = ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL)) < 0) {
rt->client_report_size = 1048576;
rt->bytes_read = 0;
rt->last_bytes_read = 0;
+ rt->server_bw = 2500000;
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
proto, path, rt->app, rt->playpath);
int pktsize, pkttype;
uint32_t ts;
const uint8_t *buf_temp = buf;
+ uint8_t c;
int ret;
do {
rt->flv_size = 0;
rt->flv_off = 0;
rt->flv_header_bytes = 0;
+ rt->flv_nb_packets++;
}
} while (buf_temp - buf < size);
+
+ if (rt->flv_nb_packets < rt->flush_interval)
+ return size;
+ rt->flv_nb_packets = 0;
+
+ /* set stream into nonblocking mode */
+ rt->stream->flags |= AVIO_FLAG_NONBLOCK;
+
+ /* try to read one byte from the stream */
+ ret = ffurl_read(rt->stream, &c, 1);
+
+ /* switch the stream back into blocking mode */
+ rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
+
+ if (ret == AVERROR(EAGAIN)) {
+ /* no incoming data to handle */
+ return size;
+ } else if (ret < 0) {
+ return ret;
+ } else if (ret == 1) {
+ RTMPPacket rpkt = { 0 };
+
+ if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
+ rt->chunk_size,
+ rt->prev_pkt[0], c)) <= 0)
+ return ret;
+
+ if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
+ return ret;
+
+ ff_rtmp_packet_destroy(&rpkt);
+ }
+
return size;
}
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_buffer", "Set buffer time in milliseconds. The default is 3000.", OFFSET(client_buffer_time), AV_OPT_TYPE_INT, {3000}, 0, INT_MAX, DEC|ENC},
{"rtmp_conn", "Append arbitrary AMF data to the Connect message", OFFSET(conn), 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_flush_interval", "Number of packets flushed in the same request (RTMPT only).", OFFSET(flush_interval), AV_OPT_TYPE_INT, {10}, 0, INT_MAX, 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"},
.flags = URL_PROTOCOL_FLAG_NETWORK,
.priv_data_class= &rtmp_class,
};
+
+static const AVClass rtmpt_class = {
+ .class_name = "rtmpt",
+ .item_name = av_default_item_name,
+ .option = rtmp_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+URLProtocol ff_rtmpt_protocol = {
+ .name = "rtmpt",
+ .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 = &rtmpt_class,
+};
projects / ffmpeg / blobdiff