int connect_oc(srs_rtmp_t ortmp) {
int ret = 0;
if ((ret = srs_rtmp_handshake(ortmp)) != 0) {
srs_human_trace("ortmp simple handshake failed. ret=%d", ret);
return ret;
}
srs_human_trace("ortmp simple handshake success");
if ((ret = srs_rtmp_connect_app(ortmp)) != 0) {
srs_human_trace("ortmp connect vhost/app failed. ret=%d", ret);
return ret;
}
srs_human_trace("ortmp connect vhost/app success");
if ((ret = srs_rtmp_publish_stream(ortmp)) != 0) {
srs_human_trace("ortmp publish stream failed. ret=%d", ret);
return ret;
}
srs_human_trace("ortmp publish stream success");
return ret;
}
int main(int argc, char** argv)
{
int ret = 0;
srs_rtmp_t rtmp;
// packet data
int size;
char type;
char* data;
u_int32_t timestamp;
// srs debug info.
char srs_server_ip[128];
char srs_server[128];
char srs_primary_authors[128];
char srs_version[32];
int srs_id = 0;
int srs_pid = 0;
// bandwidth test data.
int64_t start_time = 0;
int64_t end_time = 0;
int play_kbps = 0;
int publish_kbps = 0;
int play_bytes = 0;
int publish_bytes = 0;
int play_duration = 0;
int publish_duration = 0;
// set to zero.
srs_server_ip[0] = 0;
srs_server[0] = 0;
srs_primary_authors[0] = 0;
srs_version[0] = 0;
printf("RTMP bandwidth check/test with server.\n");
printf("srs(simple-rtmp-server) client librtmp library.\n");
printf("version: %d.%d.%d\n", srs_version_major(), srs_version_minor(), srs_version_revision());
if (argc <= 1) {
printf("RTMP bandwidth check/test with server.\n"
"Usage: %s <rtmp_url>\n"
" rtmp_url RTMP bandwidth url to check. format: rtmp://server:port/app?key=xxx,vhost=xxx\n"
"For example:\n"
" %s rtmp://127.0.0.1:1935/app?key=35c9b402c12a7246868752e2878f7e0e,vhost=bandcheck.srs.com\n"
" %s rtmp://127.0.0.1:1935/app?key=35c9b402c12a7246868752e2878f7e0e,vhost=bandcheck.srs.com>/dev/null\n"
"@remark, output text to stdout, while json to stderr.\n",
argv[0], argv[0], argv[0]);
exit(-1);
}
rtmp = srs_rtmp_create2(argv[1]);
srs_human_trace("bandwidth check/test url: %s", argv[1]);
if ((ret = srs_rtmp_handshake(rtmp)) != 0) {
srs_human_trace("simple handshake failed.");
goto rtmp_destroy;
}
srs_human_trace("simple handshake success");
if ((ret = srs_rtmp_connect_app2(rtmp,
srs_server_ip, srs_server, srs_primary_authors, srs_version, &srs_id, &srs_pid)) != 0) {
srs_human_trace("connect vhost/app failed.");
goto rtmp_destroy;
}
srs_human_trace("connect vhost/app success");
if ((ret = srs_rtmp_bandwidth_check(rtmp,
&start_time, &end_time, &play_kbps, &publish_kbps,
&play_bytes, &publish_bytes, &play_duration, &publish_duration)) != 0
) {
srs_human_trace("bandwidth check/test failed.");
goto rtmp_destroy;
}
srs_human_trace("bandwidth check/test success");
srs_human_trace("\n%s, %s\n"
"%s, %s, srs_pid=%d, srs_id=%d\n"
"duration: %dms(%d+%d)\n"
"play: %dkbps\n"
"publish: %dkbps",
(char*)srs_server, (char*)srs_primary_authors,
(char*)srs_server_ip, (char*)srs_version, srs_pid, srs_id,
(int)(end_time - start_time), play_duration, publish_duration,
play_kbps,
publish_kbps);
rtmp_destroy:
srs_rtmp_destroy(rtmp);
fprintf(stderr, "{\"code\":%d,"
"\"srs_server\":\"%s\", "
"\"srs_primary_authors\":\"%s\", "
"\"srs_server_ip\":\"%s\", "
"\"srs_version\":\"%s\", "
"\"srs_pid\":%d, "
"\"srs_id\":%d, "
"\"duration\":%d, "
"\"play_duration\":%d, "
"\"play_kbps\":%d, "
"\"publish_kbps\":%d"
"}",
ret,
(char*)srs_server, (char*)srs_primary_authors,
(char*)srs_server_ip, (char*)srs_version, srs_pid, srs_id,
(int)(end_time - start_time), play_duration, publish_duration,
play_kbps, publish_kbps);
srs_human_trace("");
srs_human_trace("completed");
return ret;
}
int main(int argc, char** argv)
{
printf("publish raw h.264 as rtmp stream to server like FMLE/FFMPEG/Encoder\n");
printf("SRS(simple-rtmp-server) client librtmp library.\n");
printf("version: %d.%d.%d\n", srs_version_major(), srs_version_minor(), srs_version_revision());
if (argc <= 2) {
printf("Usage: %s <h264_raw_file> <rtmp_publish_url>\n", argv[0]);
printf(" h264_raw_file: the h264 raw steam file.\n");
printf(" rtmp_publish_url: the rtmp publish url.\n");
printf("For example:\n");
printf(" %s ./720p.h264.raw rtmp://127.0.0.1:1935/live/livestream\n", argv[0]);
printf("Where the file: http://winlinvip.github.io/srs.release/3rdparty/720p.h264.raw\n");
printf("See: https://github.com/simple-rtmp-server/srs/issues/66\n");
exit(-1);
}
const char* raw_file = argv[1];
const char* rtmp_url = argv[2];
srs_human_trace("raw_file=%s, rtmp_url=%s", raw_file, rtmp_url);
// open file
int raw_fd = open(raw_file, O_RDONLY);
if (raw_fd < 0) {
srs_human_trace("open h264 raw file %s failed.", raw_file);
goto rtmp_destroy;
}
off_t file_size = lseek(raw_fd, 0, SEEK_END);
if (file_size <= 0) {
srs_human_trace("h264 raw file %s empty.", raw_file);
goto rtmp_destroy;
}
srs_human_trace("read entirely h264 raw file, size=%dKB", (int)(file_size / 1024));
char* h264_raw = (char*)malloc(file_size);
if (!h264_raw) {
srs_human_trace("alloc raw buffer failed for file %s.", raw_file);
goto rtmp_destroy;
}
lseek(raw_fd, 0, SEEK_SET);
ssize_t nb_read = 0;
if ((nb_read = read(raw_fd, h264_raw, file_size)) != file_size) {
srs_human_trace("buffer %s failed, expect=%dKB, actual=%dKB.",
raw_file, (int)(file_size / 1024), (int)(nb_read / 1024));
goto rtmp_destroy;
}
// connect rtmp context
srs_rtmp_t rtmp = srs_rtmp_create(rtmp_url);
if (srs_rtmp_handshake(rtmp) != 0) {
srs_human_trace("simple handshake failed.");
goto rtmp_destroy;
}
srs_human_trace("simple handshake success");
if (srs_rtmp_connect_app(rtmp) != 0) {
srs_human_trace("connect vhost/app failed.");
goto rtmp_destroy;
}
srs_human_trace("connect vhost/app success");
if (srs_rtmp_publish_stream(rtmp) != 0) {
srs_human_trace("publish stream failed.");
goto rtmp_destroy;
}
srs_human_trace("publish stream success");
int dts = 0;
int pts = 0;
// @remark, the dts and pts if read from device, for instance, the encode lib,
// so we assume the fps is 25, and each h264 frame is 1000ms/25fps=40ms/f.
int fps = 25;
// @remark, to decode the file.
char* p = h264_raw;
for (;p < h264_raw + file_size;) {
// @remark, read a frame from file buffer.
char* data = NULL;
int size = 0;
int nb_start_code = 0;
if (read_h264_frame(h264_raw, (int)file_size, &p, &nb_start_code, fps, &data, &size, &dts, &pts) < 0) {
srs_human_trace("read a frame from file buffer failed.");
goto rtmp_destroy;
}
// send out the h264 packet over RTMP
int ret = srs_h264_write_raw_frames(rtmp, data, size, dts, pts);
if (ret != 0) {
if (srs_h264_is_dvbsp_error(ret)) {
srs_human_trace("ignore drop video error, code=%d", ret);
} else if (srs_h264_is_duplicated_sps_error(ret)) {
srs_human_trace("ignore duplicated sps, code=%d", ret);
} else if (srs_h264_is_duplicated_pps_error(ret)) {
srs_human_trace("ignore duplicated pps, code=%d", ret);
} else {
srs_human_trace("send h264 raw data failed. ret=%d", ret);
goto rtmp_destroy;
}
}
// 5bits, 7.3.1 NAL unit syntax,
// H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
u_int8_t nut = (char)data[nb_start_code] & 0x1f;
srs_human_trace("sent packet: type=%s, time=%d, size=%d, fps=%d, b[%d]=%#x(%s)",
srs_human_flv_tag_type2string(SRS_RTMP_TYPE_VIDEO), dts, size, fps, nb_start_code, (char)data[nb_start_code],
(nut == 7? "SPS":(nut == 8? "PPS":(nut == 5? "I":(nut == 1? "P":"Unknown")))));
// @remark, when use encode device, it not need to sleep.
usleep(1000 / fps * 1000);
}
srs_human_trace("h264 raw data completed");
rtmp_destroy:
srs_rtmp_destroy(rtmp);
close(raw_fd);
free(h264_raw);
return 0;
}
int main(int argc, char** argv)
{
printf("publish raw audio as rtmp stream to server like FMLE/FFMPEG/Encoder\n");
printf("SRS(simple-rtmp-server) client librtmp library.\n");
printf("version: %d.%d.%d\n", srs_version_major(), srs_version_minor(), srs_version_revision());
if (argc <= 2) {
printf("Usage: %s <audio_raw_file> <rtmp_publish_url>\n", argv[0]);
printf(" audio_raw_file: the audio raw steam file.\n");
printf(" rtmp_publish_url: the rtmp publish url.\n");
printf("For example:\n");
printf(" %s ./audio.raw.pcm rtmp://127.0.0.1:1935/live/livestream\n", argv[0]);
printf("Where the file: http://winlinvip.github.io/srs.release/3rdparty/audio.raw.pcm\n");
printf("See: https://github.com/winlinvip/simple-rtmp-server/issues/212\n");
exit(-1);
}
const char* raw_file = argv[1];
const char* rtmp_url = argv[2];
srs_human_trace("raw_file=%s, rtmp_url=%s", raw_file, rtmp_url);
// open file
int raw_fd = open(raw_file, O_RDONLY);
if (raw_fd < 0) {
srs_human_trace("open audio raw file %s failed.", raw_file);
goto rtmp_destroy;
}
off_t file_size = lseek(raw_fd, 0, SEEK_END);
if (file_size <= 0) {
srs_human_trace("audio raw file %s empty.", raw_file);
goto rtmp_destroy;
}
srs_human_trace("read entirely audio raw file, size=%dKB", (int)(file_size / 1024));
char* audio_raw = (char*)malloc(file_size);
if (!audio_raw) {
srs_human_trace("alloc raw buffer failed for file %s.", raw_file);
goto rtmp_destroy;
}
lseek(raw_fd, 0, SEEK_SET);
ssize_t nb_read = 0;
if ((nb_read = read(raw_fd, audio_raw, file_size)) != file_size) {
srs_human_trace("buffer %s failed, expect=%dKB, actual=%dKB.",
raw_file, (int)(file_size / 1024), (int)(nb_read / 1024));
goto rtmp_destroy;
}
// connect rtmp context
srs_rtmp_t rtmp = srs_rtmp_create(rtmp_url);
if (srs_rtmp_handshake(rtmp) != 0) {
srs_human_trace("simple handshake failed.");
goto rtmp_destroy;
}
srs_human_trace("simple handshake success");
if (srs_rtmp_connect_app(rtmp) != 0) {
srs_human_trace("connect vhost/app failed.");
goto rtmp_destroy;
}
srs_human_trace("connect vhost/app success");
if (srs_rtmp_publish_stream(rtmp) != 0) {
srs_human_trace("publish stream failed.");
goto rtmp_destroy;
}
srs_human_trace("publish stream success");
u_int32_t timestamp = 0;
u_int32_t time_delta = 17;
// @remark, to decode the file.
char* p = audio_raw;
for (;p < audio_raw + file_size;) {
// @remark, read a frame from file buffer.
char* data = NULL;
int size = 0;
if (read_audio_frame(audio_raw, file_size, &p, &data, &size) < 0) {
srs_human_trace("read a frame from file buffer failed.");
goto rtmp_destroy;
}
// 0 = Linear PCM, platform endian
// 1 = ADPCM
// 2 = MP3
// 7 = G.711 A-law logarithmic PCM
// 8 = G.711 mu-law logarithmic PCM
// 10 = AAC
// 11 = Speex
char sound_format = 1;
// 3 = 44 kHz
char sound_rate = 3;
// 1 = 16-bit samples
char sound_size = 1;
// 1 = Stereo sound
char sound_type = 1;
timestamp += time_delta;
if (srs_audio_write_raw_frame(rtmp,
sound_format, sound_rate, sound_size, sound_type,
data, size, timestamp) != 0
) {
srs_human_trace("send audio raw data failed.");
goto rtmp_destroy;
}
srs_human_trace("sent packet: type=%s, time=%d, size=%d, codec=%d, rate=%d, sample=%d, channel=%d",
srs_human_flv_tag_type2string(SRS_RTMP_TYPE_AUDIO), timestamp, size, sound_format, sound_rate, sound_size,
sound_type);
// @remark, when use encode device, it not need to sleep.
usleep(1000 * time_delta);
}
rtmp_destroy:
srs_rtmp_destroy(rtmp);
close(raw_fd);
free(audio_raw);
return 0;
}
void PushEngine::run(){
while(1){
int64_t currentTime = srs_utils_time_ms();
if(currentTime - m_timeTick > 1000){
m_timeTick = currentTime;
#if 0
for(std::list<OneStream*>::iterator it = m_streamList.begin(); it != m_streamList.end(); it++){
m_logger.information("==> current publish [rtmp://%s/live/%s%d] size[%d]", (*it)->ipPort, \
(*it)->prefixName, (*it)->randNum, (*it)->receiveSize/1000);
(*it)->receiveSize = 0;
}
#endif
}
if(m_jobQueue.size() > 0){
OneStreamSharePtr node = m_jobQueue.get();
if(node.get() != NULL){
node->setExpiredTime(currentTime);
node->m_sendIndex = 0;
m_streamList.push_back(node);
}
std::string url = node->getURL();
node->rtmp = srs_rtmp_create(url.c_str());
m_logger.information("begin publish to [%s].", url);
if (srs_rtmp_handshake(node->rtmp) != 0) {
srs_rtmp_destroy(node->rtmp); // 析构rtmp
m_streamEventQueue.put(StreamEventSharePtr(new StreamEvent(node, currentTime, HANDSHAKE_FAIL))); // 发送握手失败事件
m_logger.error("simple handshake to [%s] failed.", url);
}
if (srs_rtmp_connect_app(node->rtmp) != 0) {
srs_rtmp_destroy(node->rtmp); // 析构rtmp
m_streamEventQueue.put(StreamEventSharePtr(new StreamEvent(node, currentTime, CONNECT_FAIL))); // 发送连接失败事件
m_logger.error("connect [%s] vhost/app failed.", url);
continue;
}
if (srs_rtmp_publish_stream(node->rtmp) != 0) {
srs_rtmp_destroy(node->rtmp); // 析构rtmp
m_streamEventQueue.put(StreamEventSharePtr(new StreamEvent(node, currentTime, PUBLIC_FAIL))); // 发送public失败事件
m_logger.error("publish stream [%s] failed.", url);
continue;
}
m_streamEventQueue.put(StreamEventSharePtr(new StreamEvent(node, currentTime, START_STREAM))); // 发送开始推流事件
}
// 发送当前时间片的帧到所有链接
for(std::list<OneStreamSharePtr>::iterator it = m_streamList.begin(); it != m_streamList.end(); ){
bool endThisStream = false;
if((*it)->m_sendIndex == 0 || (*it)->m_sendIndex + 1 > m_flvFrame.size()){
(*it)->m_baseTimestamp = currentTime;
(*it)->m_sendIndex = 0;
}
if(((*it)->m_baseTimestamp + m_flvFrame[(*it)->m_sendIndex].timestamp) <= currentTime){
Frame f = m_flvFrame[((*it)->m_sendIndex)++];
char* data = (char*)malloc(f.size);
memcpy(data, f.data, f.size);
if (srs_rtmp_write_packet((*it)->rtmp, f.type, ((*it)->m_baseTimestamp + f.timestamp) % 10000000, data, f.size) != 0) {
endThisStream = true;
m_streamEventQueue.put(StreamEventSharePtr(new StreamEvent((*it), currentTime, PUSH_STREAM_FAIL))); // 发送推流失败事件
m_logger.error("error to publish [%s].", (*it)->getURL());
}
(*it)->addReceiveSize(f.size);
}
// 推流时间到
if(currentTime > (*it)->getExpiredTime()){
endThisStream = true;
m_streamEventQueue.put(StreamEventSharePtr(new StreamEvent((*it), currentTime, END_STREAM))); // 发送推流结束事件
m_logger.information("stop to publish [%s].", (*it)->getURL());
}
if(endThisStream){
srs_rtmp_destroy((*it)->rtmp); // 析构rtmp
it = m_streamList.erase(it); // 从链表中删除
continue;
}
else{
it++;
}
}
usleep(10 * 1000);
}
}
int main(int argc, char **argv) {
printf("publish rtmp stream to server like FMLE/FFMPEG/Encoder\n");
printf("srs(simple-rtmp-server) client librtmp library.\n");
printf("version: %d.%d.%d\n", srs_version_major(), srs_version_minor(), srs_version_revision());
if (argc <= 1) {
printf("Usage: %s <rtmp_url>\n"
" rtmp_url RTMP stream url to publish\n"
"For example:\n"
" %s rtmp://127.0.0.1:1935/live/livestream\n",
argv[0], argv[0]);
exit(-1);
}
// warn it .
// @see: https://github.com/winlinvip/simple-rtmp-server/issues/126
srs_human_trace("\033[33m%s\033[0m",
"[warning] it's only a sample to use librtmp. "
"please never use it to publish and test forward/transcode/edge/HLS whatever. "
"you should refer to this tool to use the srs-librtmp to publish the real media stream."
"read about: https://github.com/winlinvip/simple-rtmp-server/issues/126");
srs_human_trace("rtmp url: %s", argv[1]);
srs_rtmp_t rtmp = srs_rtmp_create(argv[1]);
if (srs_rtmp_handshake(rtmp) != 0) {
srs_human_trace("simple handshake failed.");
goto rtmp_destroy;
}
srs_human_trace("simple handshake success");
if (srs_rtmp_connect_app(rtmp) != 0) {
srs_human_trace("connect vhost/app failed.");
goto rtmp_destroy;
}
srs_human_trace("connect vhost/app success");
if (srs_rtmp_publish_stream(rtmp) != 0) {
srs_human_trace("publish stream failed.");
goto rtmp_destroy;
}
srs_human_trace("publish stream success");
u_int32_t timestamp = 0;
for (; ;) {
char type = SRS_RTMP_TYPE_VIDEO;
int size = 4096;
char *data = (char *) malloc(4096);
timestamp += 40;
if (srs_rtmp_write_packet(rtmp, type, timestamp, data, size) != 0) {
goto rtmp_destroy;
}
srs_human_trace("sent packet: type=%s, time=%d, size=%d",
srs_human_flv_tag_type2string(type), timestamp, size);
usleep(40 * 1000);
}
rtmp_destroy:
srs_rtmp_destroy(rtmp);
return 0;
}
