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merge from 2.0release

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winlin 2015-07-14 11:32:40 +08:00
commit 5f91fbc970
10 changed files with 97 additions and 34 deletions
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7
README.md
View file

@ -1,13 +1,13 @@
#Simple-RTMP-Server
SRS/3.0，开发代号：[OuXuli](https://github.com/simple-rtmp-server/srs/wiki/v1_CN_Product#release30)
SRS/3.0, [OuXuli](https://github.com/simple-rtmp-server/srs/wiki/v1_CN_Product#release30)
SRS定位是运营级的互联网直播服务器集群，追求更好的概念完整性和最简单实现的代码。<br/>
SRS is industrial-strength live streaming cluster, for the best conceptual integrity and the simplest implementation.
Download from github.io: [Centos6-x86_64][centos0], [more...][more0]<br/>
Download from ossrs.net: [Centos6-x86_64][centos1], [more...][more1]<br/>
Contact by QQ or Skype, read [Contact][contact]
Website for SRS/2.0, read SRS 2.0 [Chinese][srs2_CN] or [English][srs2_EN].
## Why SRS?
@ -1025,5 +1025,6 @@ Winlin
[centos0]: http://winlinvip.github.io/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.32.zip
[centos1]: http://www.ossrs.net/srs.release/releases/files/SRS-CentOS6-x86_64-1.0.32.zip
[srs2_CN]: https://github.com/simple-rtmp-server/srs/wiki/v2_CN_Home
[srs2_EN]: https://github.com/simple-rtmp-server/srs/wiki/v2_EN_Home

3
trunk/research/librtmp/srs_ingest_flv.c
View file

@ -38,6 +38,7 @@ int proxy(srs_flv_t flv, srs_rtmp_t ortmp);
int connect_oc(srs_rtmp_t ortmp);
#define RE_PULSE_MS 300
#define RE_PULSE_JITTER_MS 3000
int64_t re_create();
void re_update(int64_t re, int32_t starttime, u_int32_t time);
void re_cleanup(int64_t re, int32_t starttime, u_int32_t time);
@ -256,7 +257,7 @@ void re_update(int64_t re, int32_t starttime, u_int32_t time)
// send by pulse algorithm.
int64_t now = srs_utils_time_ms();
int64_t diff = time - starttime - (now -re);
if (diff > RE_PULSE_MS) {
if (diff > RE_PULSE_MS && diff < RE_PULSE_JITTER_MS) {
usleep((useconds_t)(diff * 1000));
}
}

16
trunk/src/app/srs_app_config.cpp
View file

@ -853,6 +853,7 @@ int SrsConfig::reload_vhost(SrsConfDirective* old_root)
}
}
// TODO: reload new http_remux in on_vhost_add
// http_remux, only one per vhost.
if (get_vhost_http_remux_enabled(vhost)) {
for (it = subscribes.begin(); it != subscribes.end(); ++it) {
@ -4417,3 +4418,18 @@ bool srs_config_dvr_is_plan_append(string plan)
{
return plan == SRS_CONF_DEFAULT_DVR_PLAN_APPEND;
}
bool srs_stream_caster_is_udp(string caster)
{
return caster == SRS_CONF_DEFAULT_STREAM_CASTER_MPEGTS_OVER_UDP;
}
bool srs_stream_caster_is_rtsp(string caster)
{
return caster == SRS_CONF_DEFAULT_STREAM_CASTER_RTSP;
}
bool srs_stream_caster_is_flv(string caster)
{
return caster == SRS_CONF_DEFAULT_STREAM_CASTER_FLV;
}

3
trunk/src/app/srs_app_config.hpp
View file

@ -1143,6 +1143,9 @@ extern bool srs_config_ingest_is_stream(std::string type);
extern bool srs_config_dvr_is_plan_segment(std::string plan);
extern bool srs_config_dvr_is_plan_session(std::string plan);
extern bool srs_config_dvr_is_plan_append(std::string plan);
extern bool srs_stream_caster_is_udp(std::string caster);
extern bool srs_stream_caster_is_rtsp(std::string caster);
extern bool srs_stream_caster_is_flv(std::string caster);
// global config
extern SrsConfig* _srs_config;

6
trunk/src/app/srs_app_forward.cpp
View file

@ -164,7 +164,7 @@ int SrsForwarder::on_meta_data(SrsSharedPtrMessage* shared_metadata)
SrsSharedPtrMessage* metadata = shared_metadata->copy();
// TODO: FIXME: config the jitter of Forwarder.
if ((ret = jitter->correct(metadata, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
if ((ret = jitter->correct(metadata, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
srs_freep(metadata);
return ret;
}
@ -183,7 +183,7 @@ int SrsForwarder::on_audio(SrsSharedPtrMessage* shared_audio)
SrsSharedPtrMessage* msg = shared_audio->copy();
// TODO: FIXME: config the jitter of Forwarder.
if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
srs_freep(msg);
return ret;
}
@ -207,7 +207,7 @@ int SrsForwarder::on_video(SrsSharedPtrMessage* shared_video)
SrsSharedPtrMessage* msg = shared_video->copy();
// TODO: FIXME: config the jitter of Forwarder.
if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
if ((ret = jitter->correct(msg, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
srs_freep(msg);
return ret;
}

42
trunk/src/app/srs_app_hls.cpp
View file

@ -53,6 +53,8 @@ using namespace std;
// drop the segment when duration of ts too small.
#define SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS 100
// when hls timestamp jump, reset it.
#define SRS_AUTO_HLS_SEGMENT_TIMESTAMP_JUMP_MS 300
// fragment plus the deviation percent.
#define SRS_HLS_FLOOR_REAP_PERCENT 0.3
@ -161,6 +163,11 @@ void SrsHlsSegment::update_duration(int64_t current_frame_dts)
// update the segment duration, which is nagetive,
// just ignore it.
if (current_frame_dts < segment_start_dts) {
// for atc and timestamp jump, reset the start dts.
if (current_frame_dts < segment_start_dts - SRS_AUTO_HLS_SEGMENT_TIMESTAMP_JUMP_MS * 90) {
srs_warn("hls timestamp jump %"PRId64"=>%"PRId64, segment_start_dts, current_frame_dts);
segment_start_dts = current_frame_dts;
}
return;
}
@ -280,7 +287,7 @@ SrsHlsMuxer::SrsHlsMuxer()
previous_floor_ts = 0;
accept_floor_ts = 0;
hls_ts_floor = false;
target_duration = 0;
max_td = 0;
_sequence_no = 0;
current = NULL;
acodec = SrsCodecAudioReserved1;
@ -400,10 +407,8 @@ int SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix,
m3u8_url = srs_path_build_stream(m3u8_file, req->vhost, req->app, req->stream);
m3u8 = path + "/" + m3u8_url;
// we always keep the target duration increasing.
int max_td = srs_max(target_duration, (int)(fragment * _srs_config->get_hls_td_ratio(r->vhost)));
srs_info("hls update target duration %d=>%d, aof=%.2f", target_duration, max_td, aof_ratio);
target_duration = max_td;
// when update config, reset the history target duration.
max_td = (int)(fragment * _srs_config->get_hls_td_ratio(r->vhost));
std::string storage = _srs_config->get_hls_storage(r->vhost);
if (storage == "ram") {
@ -699,7 +704,9 @@ int SrsHlsMuxer::segment_close(string log_desc)
srs_assert(it == segments.end());
// valid, add to segments if segment duration is ok
if (current->duration * 1000 >= SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS) {
// when too small, it maybe not enough data to play.
// when too large, it maybe timestamp corrupt.
if (current->duration * 1000 >= SRS_AUTO_HLS_SEGMENT_MIN_DURATION_MS && (int)current->duration <= max_td) {
segments.push_back(current);
// use async to call the http hooks, for it will cause thread switch.
@ -750,7 +757,6 @@ int SrsHlsMuxer::segment_close(string log_desc)
// rename from tmp to real path
std::string tmp_file = current->full_path + ".tmp";
if (should_write_file) {
unlink(tmp_file.c_str());
if (unlink(tmp_file.c_str()) < 0) {
srs_warn("ignore unlink path failed, file=%s.", tmp_file.c_str());
}
@ -817,6 +823,11 @@ int SrsHlsMuxer::refresh_m3u8()
{
int ret = ERROR_SUCCESS;
// no segments, also no m3u8, return.
if (segments.size() == 0) {
return ret;
}
std::string temp_m3u8 = m3u8 + ".temp";
if ((ret = _refresh_m3u8(temp_m3u8)) == ERROR_SUCCESS) {
if (should_write_file && rename(temp_m3u8.c_str(), m3u8.c_str()) < 0) {
@ -826,7 +837,11 @@ int SrsHlsMuxer::refresh_m3u8()
}
// remove the temp file.
unlink(temp_m3u8.c_str());
if (srs_path_exists(temp_m3u8)) {
if (unlink(temp_m3u8.c_str()) < 0) {
srs_warn("ignore remove m3u8 failed, %s", temp_m3u8.c_str());
}
}
return ret;
}
@ -861,6 +876,9 @@ int SrsHlsMuxer::_refresh_m3u8(string m3u8_file)
ss << "#EXT-X-MEDIA-SEQUENCE:" << first->sequence_no << SRS_CONSTS_LF;
srs_verbose("write m3u8 sequence success.");
// iterator shared for td generation and segemnts wrote.
std::vector<SrsHlsSegment*>::iterator it;
// #EXT-X-TARGETDURATION:4294967295\n
/**
* @see hls-m3u8-draft-pantos-http-live-streaming-12.pdf, page 25
@ -871,11 +889,13 @@ int SrsHlsMuxer::_refresh_m3u8(string m3u8_file)
* typical target duration is 10 seconds.
*/
// @see https://github.com/simple-rtmp-server/srs/issues/304#issuecomment-74000081
std::vector<SrsHlsSegment*>::iterator it;
int target_duration = 0;
for (it = segments.begin(); it != segments.end(); ++it) {
SrsHlsSegment* segment = *it;
target_duration = srs_max(target_duration, (int)ceil(segment->duration));
}
target_duration = srs_max(target_duration, max_td);
ss << "#EXT-X-TARGETDURATION:" << target_duration << SRS_CONSTS_LF;
srs_verbose("write m3u8 duration success.");
@ -1335,7 +1355,7 @@ int SrsHls::on_audio(SrsSharedPtrMessage* shared_audio)
}
// TODO: FIXME: config the jitter of HLS.
if ((ret = jitter->correct(audio, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
if ((ret = jitter->correct(audio, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
srs_error("rtmp jitter correct audio failed. ret=%d", ret);
return ret;
}
@ -1393,7 +1413,7 @@ int SrsHls::on_video(SrsSharedPtrMessage* shared_video)
}
// TODO: FIXME: config the jitter of HLS.
if ((ret = jitter->correct(video, SrsRtmpJitterAlgorithmFULL)) != ERROR_SUCCESS) {
if ((ret = jitter->correct(video, SrsRtmpJitterAlgorithmOFF)) != ERROR_SUCCESS) {
srs_error("rtmp jitter correct video failed. ret=%d", ret);
return ret;
}

2
trunk/src/app/srs_app_hls.hpp
View file

@ -233,7 +233,7 @@ private:
int64_t previous_floor_ts;
private:
int _sequence_no;
int target_duration;
int max_td;
std::string m3u8;
std::string m3u8_url;
private:

17
trunk/src/app/srs_app_http_stream.cpp
View file

@ -805,7 +805,14 @@ int SrsHttpStreamServer::http_mount(SrsSource* s, SrsRequest* r)
entry->cache = new SrsStreamCache(s, r);
entry->stream = new SrsLiveStream(s, r, entry->cache);
srs_assert(!tmpl->req);
// TODO: FIXME: maybe refine the logic of http remux service.
// if user push streams followed:
// rtmp://test.com/live/stream1
// rtmp://test.com/live/stream2
// and they will using the same template, such as: [vhost]/[app]/[stream].flv
// so, need to free last request object, otherwise, it will cause memory leak.
srs_freep(tmpl->req);
tmpl->source = s;
tmpl->req = r->copy();
@ -1170,8 +1177,8 @@ int SrsHttpStreamServer::hijack(ISrsHttpMessage* request, ISrsHttpHandler** ph)
std::string sid = r->get_stream_url();
// check if the stream is enabled.
if (sflvs.find(sid) != sflvs.end()) {
SrsLiveEntry* entry = sflvs[sid];
if (!entry->stream->entry->enabled) {
SrsLiveEntry* s_entry = sflvs[sid];
if (!s_entry->stream->entry->enabled) {
srs_error("stream is disabled, hijack failed. ret=%d", ret);
return ret;
}
@ -1228,7 +1235,9 @@ int SrsHttpStreamServer::initialize_flv_streaming()
continue;
}
initialize_flv_entry(conf->arg0());
if ((ret = initialize_flv_entry(conf->arg0())) != ERROR_SUCCESS) {
return ret;
}
}
return ret;
}

6
trunk/src/app/srs_app_server.cpp
View file

@ -1120,11 +1120,11 @@ int SrsServer::listen_stream_caster()
SrsListener* listener = NULL;
std::string caster = _srs_config->get_stream_caster_engine(stream_caster);
if (caster == SRS_CONF_DEFAULT_STREAM_CASTER_MPEGTS_OVER_UDP) {
if (srs_stream_caster_is_udp(caster)) {
listener = new SrsUdpCasterListener(this, SrsListenerMpegTsOverUdp, stream_caster);
} else if (caster == SRS_CONF_DEFAULT_STREAM_CASTER_RTSP) {
} else if (srs_stream_caster_is_rtsp(caster)) {
listener = new SrsRtspListener(this, SrsListenerRtsp, stream_caster);
} else if (caster == SRS_CONF_DEFAULT_STREAM_CASTER_FLV) {
} else if (srs_stream_caster_is_flv(caster)) {
listener = new SrsHttpFlvListener(this, SrsListenerFlv, stream_caster);
} else {
ret = ERROR_STREAM_CASTER_ENGINE;

29
trunk/src/app/srs_app_source.cpp
View file

@ -55,8 +55,8 @@ using namespace std;
// 115 packets is 3s.
#define SRS_PURE_AUDIO_GUESS_COUNT 115
// when got these videos or audios, mix ok.
#define SRS_MIX_CORRECT_MIX_AV 10
// when got these videos or audios, pure audio or video, mix ok.
#define SRS_MIX_CORRECT_PURE_AV 10
int _srs_time_jitter_string2int(std::string time_jitter)
{
@ -849,12 +849,25 @@ void SrsMixQueue::push(SrsSharedPtrMessage* msg)
SrsSharedPtrMessage* SrsMixQueue::pop()
{
// when got 10+ videos or audios, mix ok.
// when got 1 video and 1 audio, mix ok.
if (nb_videos < SRS_MIX_CORRECT_MIX_AV && nb_audios < SRS_MIX_CORRECT_MIX_AV) {
if (nb_videos < 1 || nb_audios < 1) {
return NULL;
}
bool mix_ok = false;
// pure video
if (nb_videos >= SRS_MIX_CORRECT_PURE_AV && nb_audios == 0) {
mix_ok = true;
}
// pure audio
if (nb_audios >= SRS_MIX_CORRECT_PURE_AV && nb_videos == 0) {
mix_ok = true;
}
// got 1 video and 1 audio, mix ok.
if (nb_videos >= 1 && nb_audios >= 1) {
mix_ok = true;
}
if (!mix_ok) {
return NULL;
}
// pop the first msg.