mirror of
https://github.com/ossrs/srs.git
synced 2025-03-09 15:49:59 +00:00
## Describe ##
http_remux feature support config `has_audio`, `has_video` &
`guess_has_av` prop.
282d94d7bb/trunk/src/app/srs_app_http_stream.cpp (L630-L632)
Take `http_flv` as example, `srs` can accept both RTMP streams with only
audio, only video or both audio and video streams. It is controlled by
above three properties.
But `guess_has_av` is not implemented by `http_ts`. The problem is that
if I want publish a RTMP stream with audio or video track, the
`has_audio` and `has_video`, which are default true/on, must to be
config to match the RTMP stream, otherwise the `mpegts.js` player can't
play the `http-ts` stream.
## How to reproduce ##
1. `export SRS_VHOST_HTTP_REMUX_HAS_AUDIO=on; export
SRS_VHOST_HTTP_REMUX_HAS_VIDEO=on; export
SRS_VHOST_HTTP_REMUX_GUESS_HAS_AV=on; ./objs/srs -c
conf/http.ts.live.conf`
2. publish rtmp stream without video: `ffmpeg -re -stream_loop -1 -i
srs/trunk/doc/source.200kbps.768x320.flv -vn -acodec copy -f flv
rtmp://localhost/live/livestream`
3. open chrome browser, open
`http://localhost:8080/players/srs_player.html?schema=http`, go to
`LivePlayer`, input URL: `http://localhost:8080/live/livestream.ts`,
click play.
4. the `http://localhost:8080/live/livestream.ts` can not play.
## Solution ##
Let `http-ts` support `guess_has_av`, `http-flv` already supported. The
`guess_has_av` default value is ture/on, so the `http-ts|flv` can play
any streams with audio, video or both.
---------
Co-authored-by: Winlin <winlinvip@gmail.com>
1256 lines
36 KiB
C++
Executable file
1256 lines
36 KiB
C++
Executable file
//
|
|
// Copyright (c) 2013-2024 The SRS Authors
|
|
//
|
|
// SPDX-License-Identifier: MIT
|
|
//
|
|
|
|
#include <srs_app_http_stream.hpp>
|
|
|
|
#define SRS_STREAM_CACHE_CYCLE (30 * SRS_UTIME_SECONDS)
|
|
|
|
#include <sys/types.h>
|
|
#include <sys/stat.h>
|
|
#include <fcntl.h>
|
|
#include <stdlib.h>
|
|
#include <unistd.h>
|
|
|
|
#include <sstream>
|
|
using namespace std;
|
|
|
|
#include <srs_protocol_stream.hpp>
|
|
#include <srs_protocol_utility.hpp>
|
|
#include <srs_kernel_log.hpp>
|
|
#include <srs_kernel_error.hpp>
|
|
#include <srs_app_st.hpp>
|
|
#include <srs_core_autofree.hpp>
|
|
#include <srs_app_config.hpp>
|
|
#include <srs_kernel_utility.hpp>
|
|
#include <srs_kernel_file.hpp>
|
|
#include <srs_kernel_flv.hpp>
|
|
#include <srs_protocol_rtmp_stack.hpp>
|
|
#include <srs_app_source.hpp>
|
|
#include <srs_protocol_rtmp_msg_array.hpp>
|
|
#include <srs_kernel_aac.hpp>
|
|
#include <srs_kernel_mp3.hpp>
|
|
#include <srs_kernel_ts.hpp>
|
|
#include <srs_app_pithy_print.hpp>
|
|
#include <srs_app_source.hpp>
|
|
#include <srs_app_server.hpp>
|
|
#include <srs_app_statistic.hpp>
|
|
#include <srs_app_recv_thread.hpp>
|
|
#include <srs_app_http_hooks.hpp>
|
|
|
|
SrsBufferCache::SrsBufferCache(SrsRequest* r)
|
|
{
|
|
req = r->copy()->as_http();
|
|
queue = new SrsMessageQueue(true);
|
|
trd = new SrsSTCoroutine("http-stream", this);
|
|
|
|
// TODO: FIXME: support reload.
|
|
fast_cache = _srs_config->get_vhost_http_remux_fast_cache(req->vhost);
|
|
}
|
|
|
|
SrsBufferCache::~SrsBufferCache()
|
|
{
|
|
srs_freep(trd);
|
|
|
|
srs_freep(queue);
|
|
srs_freep(req);
|
|
}
|
|
|
|
srs_error_t SrsBufferCache::update_auth(SrsRequest* r)
|
|
{
|
|
srs_freep(req);
|
|
req = r->copy();
|
|
|
|
return srs_success;
|
|
}
|
|
|
|
srs_error_t SrsBufferCache::start()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = trd->start()) != srs_success) {
|
|
return srs_error_wrap(err, "corotine");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsBufferCache::stop()
|
|
{
|
|
trd->stop();
|
|
}
|
|
|
|
bool SrsBufferCache::alive()
|
|
{
|
|
srs_error_t err = trd->pull();
|
|
if (err == srs_success) {
|
|
return true;
|
|
}
|
|
|
|
srs_freep(err);
|
|
return false;
|
|
}
|
|
|
|
srs_error_t SrsBufferCache::dump_cache(SrsLiveConsumer* consumer, SrsRtmpJitterAlgorithm jitter)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (fast_cache <= 0) {
|
|
return err;
|
|
}
|
|
|
|
// the jitter is get from SrsLiveSource, which means the time_jitter of vhost.
|
|
if ((err = queue->dump_packets(consumer, false, jitter)) != srs_success) {
|
|
return srs_error_wrap(err, "dump packets");
|
|
}
|
|
|
|
srs_trace("http: dump cache %d msgs, duration=%dms, cache=%dms",
|
|
queue->size(), srsu2msi(queue->duration()), srsu2msi(fast_cache));
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsBufferCache::cycle()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// TODO: FIXME: support reload.
|
|
if (fast_cache <= 0) {
|
|
srs_usleep(SRS_STREAM_CACHE_CYCLE);
|
|
return err;
|
|
}
|
|
|
|
SrsSharedPtr<SrsLiveSource> live_source = _srs_sources->fetch(req);
|
|
if (!live_source.get()) {
|
|
return srs_error_new(ERROR_NO_SOURCE, "no source for %s", req->get_stream_url().c_str());
|
|
}
|
|
|
|
// the stream cache will create consumer to cache stream,
|
|
// which will trigger to fetch stream from origin for edge.
|
|
SrsLiveConsumer* consumer_raw = NULL;
|
|
if ((err = live_source->create_consumer(consumer_raw)) != srs_success) {
|
|
return srs_error_wrap(err, "create consumer");
|
|
}
|
|
SrsUniquePtr<SrsLiveConsumer> consumer(consumer_raw);
|
|
|
|
if ((err = live_source->consumer_dumps(consumer.get(), false, false, true)) != srs_success) {
|
|
return srs_error_wrap(err, "dumps consumer");
|
|
}
|
|
|
|
SrsUniquePtr<SrsPithyPrint> pprint(SrsPithyPrint::create_http_stream_cache());
|
|
|
|
SrsMessageArray msgs(SRS_PERF_MW_MSGS);
|
|
|
|
// set the queue size, which used for max cache.
|
|
// TODO: FIXME: support reload.
|
|
queue->set_queue_size(fast_cache);
|
|
|
|
while (true) {
|
|
if ((err = trd->pull()) != srs_success) {
|
|
return srs_error_wrap(err, "buffer cache");
|
|
}
|
|
|
|
pprint->elapse();
|
|
|
|
// get messages from consumer.
|
|
// each msg in msgs.msgs must be free, for the SrsMessageArray never free them.
|
|
int count = 0;
|
|
if ((err = consumer->dump_packets(&msgs, count)) != srs_success) {
|
|
return srs_error_wrap(err, "consumer dump packets");
|
|
}
|
|
|
|
if (count <= 0) {
|
|
srs_info("http: sleep %dms for no msg", srsu2msi(SRS_CONSTS_RTMP_PULSE));
|
|
// directly use sleep, donot use consumer wait.
|
|
srs_usleep(SRS_CONSTS_RTMP_PULSE);
|
|
|
|
// ignore when nothing got.
|
|
continue;
|
|
}
|
|
|
|
if (pprint->can_print()) {
|
|
srs_trace("-> " SRS_CONSTS_LOG_HTTP_STREAM_CACHE " http: got %d msgs, age=%d, min=%d, mw=%d",
|
|
count, pprint->age(), SRS_PERF_MW_MIN_MSGS, srsu2msi(SRS_CONSTS_RTMP_PULSE));
|
|
}
|
|
|
|
// free the messages.
|
|
for (int i = 0; i < count; i++) {
|
|
SrsSharedPtrMessage* msg = msgs.msgs[i];
|
|
queue->enqueue(msg);
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
ISrsBufferEncoder::ISrsBufferEncoder()
|
|
{
|
|
}
|
|
|
|
ISrsBufferEncoder::~ISrsBufferEncoder()
|
|
{
|
|
}
|
|
|
|
SrsTsStreamEncoder::SrsTsStreamEncoder()
|
|
{
|
|
enc = new SrsTsTransmuxer();
|
|
}
|
|
|
|
SrsTsStreamEncoder::~SrsTsStreamEncoder()
|
|
{
|
|
srs_freep(enc);
|
|
}
|
|
|
|
srs_error_t SrsTsStreamEncoder::initialize(SrsFileWriter* w, SrsBufferCache* /*c*/)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = enc->initialize(w)) != srs_success) {
|
|
return srs_error_wrap(err, "init encoder");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsTsStreamEncoder::write_audio(int64_t timestamp, char* data, int size)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = enc->write_audio(timestamp, data, size)) != srs_success) {
|
|
return srs_error_wrap(err, "write audio");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsTsStreamEncoder::write_video(int64_t timestamp, char* data, int size)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = enc->write_video(timestamp, data, size)) != srs_success) {
|
|
return srs_error_wrap(err, "write video");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsTsStreamEncoder::write_metadata(int64_t /*timestamp*/, char* /*data*/, int /*size*/)
|
|
{
|
|
return srs_success;
|
|
}
|
|
|
|
bool SrsTsStreamEncoder::has_cache()
|
|
{
|
|
// for ts stream, use gop cache of SrsLiveSource is ok.
|
|
return false;
|
|
}
|
|
|
|
srs_error_t SrsTsStreamEncoder::dump_cache(SrsLiveConsumer* /*consumer*/, SrsRtmpJitterAlgorithm /*jitter*/)
|
|
{
|
|
// for ts stream, ignore cache.
|
|
return srs_success;
|
|
}
|
|
|
|
void SrsTsStreamEncoder::set_has_audio(bool v)
|
|
{
|
|
enc->set_has_audio(v);
|
|
}
|
|
|
|
void SrsTsStreamEncoder::set_has_video(bool v)
|
|
{
|
|
enc->set_has_video(v);
|
|
}
|
|
|
|
void SrsTsStreamEncoder::set_guess_has_av(bool v)
|
|
{
|
|
enc->set_guess_has_av(v);
|
|
}
|
|
|
|
SrsFlvStreamEncoder::SrsFlvStreamEncoder()
|
|
{
|
|
header_written = false;
|
|
enc = new SrsFlvTransmuxer();
|
|
has_audio_ = true;
|
|
has_video_ = true;
|
|
guess_has_av_ = true;
|
|
}
|
|
|
|
SrsFlvStreamEncoder::~SrsFlvStreamEncoder()
|
|
{
|
|
srs_freep(enc);
|
|
}
|
|
|
|
srs_error_t SrsFlvStreamEncoder::initialize(SrsFileWriter* w, SrsBufferCache* /*c*/)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = enc->initialize(w)) != srs_success) {
|
|
return srs_error_wrap(err, "init encoder");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsFlvStreamEncoder::write_audio(int64_t timestamp, char* data, int size)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = write_header(has_video_, has_audio_)) != srs_success) {
|
|
return srs_error_wrap(err, "write header");
|
|
}
|
|
|
|
return enc->write_audio(timestamp, data, size);
|
|
}
|
|
|
|
srs_error_t SrsFlvStreamEncoder::write_video(int64_t timestamp, char* data, int size)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = write_header(has_video_, has_audio_)) != srs_success) {
|
|
return srs_error_wrap(err, "write header");
|
|
}
|
|
|
|
return enc->write_video(timestamp, data, size);
|
|
}
|
|
|
|
srs_error_t SrsFlvStreamEncoder::write_metadata(int64_t timestamp, char* data, int size)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if ((err = write_header(has_video_, has_audio_)) != srs_success) {
|
|
return srs_error_wrap(err, "write header");
|
|
}
|
|
|
|
return enc->write_metadata(SrsFrameTypeScript, data, size);
|
|
}
|
|
|
|
void SrsFlvStreamEncoder::set_drop_if_not_match(bool v)
|
|
{
|
|
enc->set_drop_if_not_match(v);
|
|
}
|
|
|
|
void SrsFlvStreamEncoder::set_has_audio(bool v)
|
|
{
|
|
has_audio_ = v;
|
|
}
|
|
|
|
void SrsFlvStreamEncoder::set_has_video(bool v)
|
|
{
|
|
has_video_ = v;
|
|
}
|
|
|
|
void SrsFlvStreamEncoder::set_guess_has_av(bool v)
|
|
{
|
|
guess_has_av_ = v;
|
|
}
|
|
|
|
bool SrsFlvStreamEncoder::has_cache()
|
|
{
|
|
// for flv stream, use gop cache of SrsLiveSource is ok.
|
|
return false;
|
|
}
|
|
|
|
srs_error_t SrsFlvStreamEncoder::dump_cache(SrsLiveConsumer* /*consumer*/, SrsRtmpJitterAlgorithm /*jitter*/)
|
|
{
|
|
// for flv stream, ignore cache.
|
|
return srs_success;
|
|
}
|
|
|
|
srs_error_t SrsFlvStreamEncoder::write_tags(SrsSharedPtrMessage** msgs, int count)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// Ignore if no messages.
|
|
if (count <= 0) return err;
|
|
|
|
// For https://github.com/ossrs/srs/issues/939
|
|
if (!header_written) {
|
|
bool has_video = has_video_; bool has_audio = has_audio_;
|
|
|
|
// See https://github.com/ossrs/srs/issues/939#issuecomment-1351385460
|
|
if (guess_has_av_) {
|
|
int nn_video_frames = 0; int nn_audio_frames = 0;
|
|
has_audio = has_video = false;
|
|
|
|
// Note that we must iterate all messages to count the audio and video frames.
|
|
for (int i = 0; i < count; i++) {
|
|
SrsSharedPtrMessage* msg = msgs[i];
|
|
if (msg->is_video()) {
|
|
if (!SrsFlvVideo::sh(msg->payload, msg->size)) nn_video_frames++;
|
|
has_video = true;
|
|
} else if (msg->is_audio()) {
|
|
if (!SrsFlvAudio::sh(msg->payload, msg->size)) nn_audio_frames++;
|
|
has_audio = true;
|
|
}
|
|
}
|
|
|
|
// See https://github.com/ossrs/srs/issues/939#issuecomment-1348541733
|
|
if (nn_video_frames > 0 && nn_audio_frames == 0) {
|
|
if (has_audio) srs_trace("FLV: Reset has_audio for videos=%d and audios=%d", nn_video_frames, nn_audio_frames);
|
|
has_audio = false;
|
|
}
|
|
if (nn_audio_frames > 0 && nn_video_frames == 0) {
|
|
if (has_video) srs_trace("FLV: Reset has_video for videos=%d and audios=%d", nn_video_frames, nn_audio_frames);
|
|
has_video = false;
|
|
}
|
|
}
|
|
|
|
// Drop data if no A+V.
|
|
if (!has_video && !has_audio) {
|
|
return err;
|
|
}
|
|
|
|
if ((err = write_header(has_video, has_audio)) != srs_success) {
|
|
return srs_error_wrap(err, "write header");
|
|
}
|
|
}
|
|
|
|
// Write tags after header is done.
|
|
return enc->write_tags(msgs, count);
|
|
}
|
|
|
|
srs_error_t SrsFlvStreamEncoder::write_header(bool has_video, bool has_audio)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (!header_written) {
|
|
header_written = true;
|
|
|
|
if ((err = enc->write_header(has_video, has_audio)) != srs_success) {
|
|
return srs_error_wrap(err, "write header");
|
|
}
|
|
|
|
srs_trace("FLV: write header audio=%d, video=%d, dinm=%d, config=%d/%d/%d", has_audio, has_video,
|
|
enc->drop_if_not_match(), has_audio_, has_video_, guess_has_av_);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
SrsAacStreamEncoder::SrsAacStreamEncoder()
|
|
{
|
|
enc = new SrsAacTransmuxer();
|
|
cache = NULL;
|
|
}
|
|
|
|
SrsAacStreamEncoder::~SrsAacStreamEncoder()
|
|
{
|
|
srs_freep(enc);
|
|
}
|
|
|
|
srs_error_t SrsAacStreamEncoder::initialize(SrsFileWriter* w, SrsBufferCache* c)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
cache = c;
|
|
|
|
if ((err = enc->initialize(w)) != srs_success) {
|
|
return srs_error_wrap(err, "init encoder");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsAacStreamEncoder::write_audio(int64_t timestamp, char* data, int size)
|
|
{
|
|
return enc->write_audio(timestamp, data, size);
|
|
}
|
|
|
|
srs_error_t SrsAacStreamEncoder::write_video(int64_t /*timestamp*/, char* /*data*/, int /*size*/)
|
|
{
|
|
// aac ignore any flv video.
|
|
return srs_success;
|
|
}
|
|
|
|
srs_error_t SrsAacStreamEncoder::write_metadata(int64_t /*timestamp*/, char* /*data*/, int /*size*/)
|
|
{
|
|
// aac ignore any flv metadata.
|
|
return srs_success;
|
|
}
|
|
|
|
bool SrsAacStreamEncoder::has_cache()
|
|
{
|
|
return true;
|
|
}
|
|
|
|
srs_error_t SrsAacStreamEncoder::dump_cache(SrsLiveConsumer* consumer, SrsRtmpJitterAlgorithm jitter)
|
|
{
|
|
srs_assert(cache);
|
|
return cache->dump_cache(consumer, jitter);
|
|
}
|
|
|
|
SrsMp3StreamEncoder::SrsMp3StreamEncoder()
|
|
{
|
|
enc = new SrsMp3Transmuxer();
|
|
cache = NULL;
|
|
}
|
|
|
|
SrsMp3StreamEncoder::~SrsMp3StreamEncoder()
|
|
{
|
|
srs_freep(enc);
|
|
}
|
|
|
|
srs_error_t SrsMp3StreamEncoder::initialize(SrsFileWriter* w, SrsBufferCache* c)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
cache = c;
|
|
|
|
if ((err = enc->initialize(w)) != srs_success) {
|
|
return srs_error_wrap(err, "init encoder");
|
|
}
|
|
|
|
if ((err = enc->write_header()) != srs_success) {
|
|
return srs_error_wrap(err, "init encoder");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsMp3StreamEncoder::write_audio(int64_t timestamp, char* data, int size)
|
|
{
|
|
return enc->write_audio(timestamp, data, size);
|
|
}
|
|
|
|
srs_error_t SrsMp3StreamEncoder::write_video(int64_t /*timestamp*/, char* /*data*/, int /*size*/)
|
|
{
|
|
// mp3 ignore any flv video.
|
|
return srs_success;
|
|
}
|
|
|
|
srs_error_t SrsMp3StreamEncoder::write_metadata(int64_t /*timestamp*/, char* /*data*/, int /*size*/)
|
|
{
|
|
// mp3 ignore any flv metadata.
|
|
return srs_success;
|
|
}
|
|
|
|
bool SrsMp3StreamEncoder::has_cache()
|
|
{
|
|
return true;
|
|
}
|
|
|
|
srs_error_t SrsMp3StreamEncoder::dump_cache(SrsLiveConsumer* consumer, SrsRtmpJitterAlgorithm jitter)
|
|
{
|
|
srs_assert(cache);
|
|
return cache->dump_cache(consumer, jitter);
|
|
}
|
|
|
|
SrsBufferWriter::SrsBufferWriter(ISrsHttpResponseWriter* w)
|
|
{
|
|
writer = w;
|
|
}
|
|
|
|
SrsBufferWriter::~SrsBufferWriter()
|
|
{
|
|
}
|
|
|
|
srs_error_t SrsBufferWriter::open(std::string /*file*/)
|
|
{
|
|
return srs_success;
|
|
}
|
|
|
|
void SrsBufferWriter::close()
|
|
{
|
|
}
|
|
|
|
bool SrsBufferWriter::is_open()
|
|
{
|
|
return true;
|
|
}
|
|
|
|
int64_t SrsBufferWriter::tellg()
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
srs_error_t SrsBufferWriter::write(void* buf, size_t count, ssize_t* pnwrite)
|
|
{
|
|
if (pnwrite) {
|
|
*pnwrite = count;
|
|
}
|
|
return writer->write((char*)buf, (int)count);
|
|
}
|
|
|
|
srs_error_t SrsBufferWriter::writev(const iovec* iov, int iovcnt, ssize_t* pnwrite)
|
|
{
|
|
return writer->writev(iov, iovcnt, pnwrite);
|
|
}
|
|
|
|
SrsLiveStream::SrsLiveStream(SrsRequest* r, SrsBufferCache* c)
|
|
{
|
|
cache = c;
|
|
req = r->copy()->as_http();
|
|
security_ = new SrsSecurity();
|
|
alive_ = false;
|
|
}
|
|
|
|
SrsLiveStream::~SrsLiveStream()
|
|
{
|
|
srs_freep(req);
|
|
srs_freep(security_);
|
|
}
|
|
|
|
srs_error_t SrsLiveStream::update_auth(SrsRequest* r)
|
|
{
|
|
srs_freep(req);
|
|
req = r->copy()->as_http();
|
|
|
|
return srs_success;
|
|
}
|
|
|
|
srs_error_t SrsLiveStream::serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsHttpMessage* hr = dynamic_cast<SrsHttpMessage*>(r);
|
|
SrsHttpConn* hc = dynamic_cast<SrsHttpConn*>(hr->connection());
|
|
SrsHttpxConn* hxc = dynamic_cast<SrsHttpxConn*>(hc->handler());
|
|
|
|
// Note that we should enable stat for HTTP streaming client, because each HTTP streaming connection is a real
|
|
// session that should have statistics for itself.
|
|
hxc->set_enable_stat(true);
|
|
|
|
// Correct the app and stream by path, which is created from template.
|
|
// @remark Be careful that the stream has extension now, might cause identify fail.
|
|
req->stream = srs_path_basename(r->path());
|
|
|
|
// update client ip
|
|
req->ip = hc->remote_ip();
|
|
|
|
// We must do stat the client before hooks, because hooks depends on it.
|
|
SrsStatistic* stat = SrsStatistic::instance();
|
|
if ((err = stat->on_client(_srs_context->get_id().c_str(), req, hc, SrsFlvPlay)) != srs_success) {
|
|
return srs_error_wrap(err, "stat on client");
|
|
}
|
|
|
|
if ((err = security_->check(SrsFlvPlay, req->ip, req)) != srs_success) {
|
|
return srs_error_wrap(err, "flv: security check");
|
|
}
|
|
|
|
// We must do hook after stat, because depends on it.
|
|
if ((err = http_hooks_on_play(r)) != srs_success) {
|
|
return srs_error_wrap(err, "http hook");
|
|
}
|
|
|
|
alive_ = true;
|
|
err = do_serve_http(w, r);
|
|
alive_ = false;
|
|
|
|
http_hooks_on_stop(r);
|
|
|
|
return err;
|
|
}
|
|
|
|
bool SrsLiveStream::alive()
|
|
{
|
|
return alive_;
|
|
}
|
|
|
|
srs_error_t SrsLiveStream::do_serve_http(ISrsHttpResponseWriter* w, ISrsHttpMessage* r)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
string enc_desc;
|
|
ISrsBufferEncoder* enc_raw = NULL;
|
|
|
|
srs_assert(entry);
|
|
bool drop_if_not_match = _srs_config->get_vhost_http_remux_drop_if_not_match(req->vhost);
|
|
bool has_audio = _srs_config->get_vhost_http_remux_has_audio(req->vhost);
|
|
bool has_video = _srs_config->get_vhost_http_remux_has_video(req->vhost);
|
|
bool guess_has_av = _srs_config->get_vhost_http_remux_guess_has_av(req->vhost);
|
|
|
|
if (srs_string_ends_with(entry->pattern, ".flv")) {
|
|
w->header()->set_content_type("video/x-flv");
|
|
enc_desc = "FLV";
|
|
enc_raw = new SrsFlvStreamEncoder();
|
|
((SrsFlvStreamEncoder*)enc_raw)->set_drop_if_not_match(drop_if_not_match);
|
|
((SrsFlvStreamEncoder*)enc_raw)->set_has_audio(has_audio);
|
|
((SrsFlvStreamEncoder*)enc_raw)->set_has_video(has_video);
|
|
((SrsFlvStreamEncoder*)enc_raw)->set_guess_has_av(guess_has_av);
|
|
} else if (srs_string_ends_with(entry->pattern, ".aac")) {
|
|
w->header()->set_content_type("audio/x-aac");
|
|
enc_desc = "AAC";
|
|
enc_raw = new SrsAacStreamEncoder();
|
|
} else if (srs_string_ends_with(entry->pattern, ".mp3")) {
|
|
w->header()->set_content_type("audio/mpeg");
|
|
enc_desc = "MP3";
|
|
enc_raw = new SrsMp3StreamEncoder();
|
|
} else if (srs_string_ends_with(entry->pattern, ".ts")) {
|
|
w->header()->set_content_type("video/MP2T");
|
|
enc_desc = "TS";
|
|
enc_raw = new SrsTsStreamEncoder();
|
|
((SrsTsStreamEncoder*)enc_raw)->set_has_audio(has_audio);
|
|
((SrsTsStreamEncoder*)enc_raw)->set_has_video(has_video);
|
|
((SrsTsStreamEncoder*)enc_raw)->set_guess_has_av(guess_has_av);
|
|
} else {
|
|
return srs_error_new(ERROR_HTTP_LIVE_STREAM_EXT, "invalid pattern=%s", entry->pattern.c_str());
|
|
}
|
|
SrsUniquePtr<ISrsBufferEncoder> enc(enc_raw);
|
|
|
|
// Enter chunked mode, because we didn't set the content-length.
|
|
w->write_header(SRS_CONSTS_HTTP_OK);
|
|
|
|
SrsSharedPtr<SrsLiveSource> live_source = _srs_sources->fetch(req);
|
|
if (!live_source.get()) {
|
|
return srs_error_new(ERROR_NO_SOURCE, "no source for %s", req->get_stream_url().c_str());
|
|
}
|
|
|
|
// create consumer of souce, ignore gop cache, use the audio gop cache.
|
|
SrsLiveConsumer* consumer_raw = NULL;
|
|
if ((err = live_source->create_consumer(consumer_raw)) != srs_success) {
|
|
return srs_error_wrap(err, "create consumer");
|
|
}
|
|
SrsUniquePtr<SrsLiveConsumer> consumer(consumer_raw);
|
|
|
|
if ((err = live_source->consumer_dumps(consumer.get(), true, true, !enc->has_cache())) != srs_success) {
|
|
return srs_error_wrap(err, "dumps consumer");
|
|
}
|
|
|
|
SrsUniquePtr<SrsPithyPrint> pprint(SrsPithyPrint::create_http_stream());
|
|
|
|
SrsMessageArray msgs(SRS_PERF_MW_MSGS);
|
|
|
|
// Use receive thread to accept the close event to avoid FD leak.
|
|
// @see https://github.com/ossrs/srs/issues/636#issuecomment-298208427
|
|
SrsHttpMessage* hr = dynamic_cast<SrsHttpMessage*>(r);
|
|
SrsHttpConn* hc = dynamic_cast<SrsHttpConn*>(hr->connection());
|
|
|
|
// the memory writer.
|
|
SrsBufferWriter writer(w);
|
|
if ((err = enc->initialize(&writer, cache)) != srs_success) {
|
|
return srs_error_wrap(err, "init encoder");
|
|
}
|
|
|
|
// if gop cache enabled for encoder, dump to consumer.
|
|
if (enc->has_cache()) {
|
|
if ((err = enc->dump_cache(consumer.get(), live_source->jitter())) != srs_success) {
|
|
return srs_error_wrap(err, "encoder dump cache");
|
|
}
|
|
}
|
|
|
|
// Try to use fast flv encoder, remember that it maybe NULL.
|
|
SrsFlvStreamEncoder* ffe = dynamic_cast<SrsFlvStreamEncoder*>(enc.get());
|
|
|
|
// Note that the handler of hc now is hxc.
|
|
SrsHttpxConn* hxc = dynamic_cast<SrsHttpxConn*>(hc->handler());
|
|
srs_assert(hxc);
|
|
|
|
// Start a thread to receive all messages from client, then drop them.
|
|
SrsUniquePtr<SrsHttpRecvThread> trd(new SrsHttpRecvThread(hxc));
|
|
|
|
if ((err = trd->start()) != srs_success) {
|
|
return srs_error_wrap(err, "start recv thread");
|
|
}
|
|
|
|
srs_utime_t mw_sleep = _srs_config->get_mw_sleep(req->vhost);
|
|
srs_trace("FLV %s, encoder=%s, mw_sleep=%dms, cache=%d, msgs=%d, dinm=%d, guess_av=%d/%d/%d",
|
|
entry->pattern.c_str(), enc_desc.c_str(), srsu2msi(mw_sleep), enc->has_cache(), msgs.max, drop_if_not_match,
|
|
has_audio, has_video, guess_has_av);
|
|
|
|
// TODO: free and erase the disabled entry after all related connections is closed.
|
|
// TODO: FXIME: Support timeout for player, quit infinite-loop.
|
|
while (entry->enabled) {
|
|
// Whether client closed the FD.
|
|
if ((err = trd->pull()) != srs_success) {
|
|
return srs_error_wrap(err, "recv thread");
|
|
}
|
|
|
|
pprint->elapse();
|
|
|
|
// get messages from consumer.
|
|
// each msg in msgs.msgs must be free, for the SrsMessageArray never free them.
|
|
int count = 0;
|
|
if ((err = consumer->dump_packets(&msgs, count)) != srs_success) {
|
|
return srs_error_wrap(err, "consumer dump packets");
|
|
}
|
|
|
|
// TODO: FIXME: Support merged-write wait.
|
|
if (count <= 0) {
|
|
// Directly use sleep, donot use consumer wait, because we couldn't awake consumer.
|
|
srs_usleep(mw_sleep);
|
|
// ignore when nothing got.
|
|
continue;
|
|
}
|
|
|
|
if (pprint->can_print()) {
|
|
srs_trace("-> " SRS_CONSTS_LOG_HTTP_STREAM " http: got %d msgs, age=%d, min=%d, mw=%d",
|
|
count, pprint->age(), SRS_PERF_MW_MIN_MSGS, srsu2msi(mw_sleep));
|
|
}
|
|
|
|
// sendout all messages.
|
|
if (ffe) {
|
|
err = ffe->write_tags(msgs.msgs, count);
|
|
} else {
|
|
err = streaming_send_messages(enc.get(), msgs.msgs, count);
|
|
}
|
|
|
|
// TODO: FIXME: Update the stat.
|
|
|
|
// free the messages.
|
|
for (int i = 0; i < count; i++) {
|
|
SrsSharedPtrMessage* msg = msgs.msgs[i];
|
|
srs_freep(msg);
|
|
}
|
|
|
|
// check send error code.
|
|
if (err != srs_success) {
|
|
return srs_error_wrap(err, "send messages");
|
|
}
|
|
}
|
|
|
|
// Here, the entry is disabled by encoder un-publishing or reloading,
|
|
// so we must return a io.EOF error to disconnect the client, or the client will never quit.
|
|
return srs_error_new(ERROR_HTTP_STREAM_EOF, "Stream EOF");
|
|
}
|
|
|
|
srs_error_t SrsLiveStream::http_hooks_on_play(ISrsHttpMessage* r)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost)) {
|
|
return err;
|
|
}
|
|
|
|
// Create request to report for the specified connection.
|
|
SrsHttpMessage* hr = dynamic_cast<SrsHttpMessage*>(r);
|
|
SrsUniquePtr<SrsRequest> nreq(hr->to_request(req->vhost));
|
|
|
|
// the http hooks will cause context switch,
|
|
// so we must copy all hooks for the on_connect may freed.
|
|
// @see https://github.com/ossrs/srs/issues/475
|
|
vector<string> hooks;
|
|
|
|
if (true) {
|
|
SrsConfDirective* conf = _srs_config->get_vhost_on_play(nreq->vhost);
|
|
|
|
if (!conf) {
|
|
return err;
|
|
}
|
|
|
|
hooks = conf->args;
|
|
}
|
|
|
|
for (int i = 0; i < (int)hooks.size(); i++) {
|
|
std::string url = hooks.at(i);
|
|
if ((err = SrsHttpHooks::on_play(url, nreq.get())) != srs_success) {
|
|
return srs_error_wrap(err, "http on_play %s", url.c_str());
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsLiveStream::http_hooks_on_stop(ISrsHttpMessage* r)
|
|
{
|
|
if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost)) {
|
|
return;
|
|
}
|
|
|
|
// Create request to report for the specified connection.
|
|
SrsHttpMessage* hr = dynamic_cast<SrsHttpMessage*>(r);
|
|
SrsUniquePtr<SrsRequest> nreq(hr->to_request(req->vhost));
|
|
|
|
// the http hooks will cause context switch,
|
|
// so we must copy all hooks for the on_connect may freed.
|
|
// @see https://github.com/ossrs/srs/issues/475
|
|
vector<string> hooks;
|
|
|
|
if (true) {
|
|
SrsConfDirective* conf = _srs_config->get_vhost_on_stop(nreq->vhost);
|
|
|
|
if (!conf) {
|
|
srs_info("ignore the empty http callback: on_stop");
|
|
return;
|
|
}
|
|
|
|
hooks = conf->args;
|
|
}
|
|
|
|
for (int i = 0; i < (int)hooks.size(); i++) {
|
|
std::string url = hooks.at(i);
|
|
SrsHttpHooks::on_stop(url, nreq.get());
|
|
}
|
|
|
|
return;
|
|
}
|
|
|
|
srs_error_t SrsLiveStream::streaming_send_messages(ISrsBufferEncoder* enc, SrsSharedPtrMessage** msgs, int nb_msgs)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// TODO: In gop cache, we know both the audio and video codec, so we should notice the encoder, which might depends
|
|
// on setting the correct codec information, for example, HTTP-TS or HLS will write PMT.
|
|
for (int i = 0; i < nb_msgs; i++) {
|
|
SrsSharedPtrMessage* msg = msgs[i];
|
|
|
|
if (msg->is_audio()) {
|
|
err = enc->write_audio(msg->timestamp, msg->payload, msg->size);
|
|
} else if (msg->is_video()) {
|
|
err = enc->write_video(msg->timestamp, msg->payload, msg->size);
|
|
} else {
|
|
err = enc->write_metadata(msg->timestamp, msg->payload, msg->size);
|
|
}
|
|
|
|
if (err != srs_success) {
|
|
return srs_error_wrap(err, "send messages");
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
SrsLiveEntry::SrsLiveEntry(std::string m)
|
|
{
|
|
mount = m;
|
|
|
|
stream = NULL;
|
|
cache = NULL;
|
|
|
|
req = NULL;
|
|
|
|
std::string ext = srs_path_filext(m);
|
|
_is_flv = (ext == ".flv");
|
|
_is_ts = (ext == ".ts");
|
|
_is_mp3 = (ext == ".mp3");
|
|
_is_aac = (ext == ".aac");
|
|
}
|
|
|
|
SrsLiveEntry::~SrsLiveEntry()
|
|
{
|
|
srs_freep(req);
|
|
}
|
|
|
|
bool SrsLiveEntry::is_flv()
|
|
{
|
|
return _is_flv;
|
|
}
|
|
|
|
bool SrsLiveEntry::is_ts()
|
|
{
|
|
return _is_ts;
|
|
}
|
|
|
|
bool SrsLiveEntry::is_aac()
|
|
{
|
|
return _is_aac;
|
|
}
|
|
|
|
bool SrsLiveEntry::is_mp3()
|
|
{
|
|
return _is_mp3;
|
|
}
|
|
|
|
SrsHttpStreamServer::SrsHttpStreamServer(SrsServer* svr)
|
|
{
|
|
server = svr;
|
|
|
|
mux.hijack(this);
|
|
_srs_config->subscribe(this);
|
|
}
|
|
|
|
SrsHttpStreamServer::~SrsHttpStreamServer()
|
|
{
|
|
mux.unhijack(this);
|
|
_srs_config->unsubscribe(this);
|
|
|
|
if (true) {
|
|
std::map<std::string, SrsLiveEntry*>::iterator it;
|
|
for (it = templateHandlers.begin(); it != templateHandlers.end(); ++it) {
|
|
SrsLiveEntry* entry = it->second;
|
|
srs_freep(entry);
|
|
}
|
|
templateHandlers.clear();
|
|
}
|
|
if (true) {
|
|
std::map<std::string, SrsLiveEntry*>::iterator it;
|
|
for (it = streamHandlers.begin(); it != streamHandlers.end(); ++it) {
|
|
SrsLiveEntry* entry = it->second;
|
|
srs_freep(entry);
|
|
}
|
|
streamHandlers.clear();
|
|
}
|
|
}
|
|
|
|
srs_error_t SrsHttpStreamServer::initialize()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// remux rtmp to flv live streaming
|
|
if ((err = initialize_flv_streaming()) != srs_success) {
|
|
return srs_error_wrap(err, "http flv stream");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
// TODO: FIXME: rename for HTTP FLV mount.
|
|
srs_error_t SrsHttpStreamServer::http_mount(SrsRequest* r)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// the id to identify stream.
|
|
std::string sid = r->get_stream_url();
|
|
SrsLiveEntry* entry = NULL;
|
|
|
|
// create stream from template when not found.
|
|
if (streamHandlers.find(sid) == streamHandlers.end()) {
|
|
if (templateHandlers.find(r->vhost) == templateHandlers.end()) {
|
|
return err;
|
|
}
|
|
|
|
SrsLiveEntry* tmpl = templateHandlers[r->vhost];
|
|
|
|
std::string mount = tmpl->mount;
|
|
|
|
// replace the vhost variable
|
|
mount = srs_string_replace(mount, "[vhost]", r->vhost);
|
|
mount = srs_string_replace(mount, "[app]", r->app);
|
|
mount = srs_string_replace(mount, "[stream]", r->stream);
|
|
|
|
// remove the default vhost mount
|
|
mount = srs_string_replace(mount, SRS_CONSTS_RTMP_DEFAULT_VHOST"/", "/");
|
|
|
|
entry = new SrsLiveEntry(mount);
|
|
|
|
entry->req = r->copy()->as_http();
|
|
entry->cache = new SrsBufferCache(r);
|
|
entry->stream = new SrsLiveStream(r, entry->cache);
|
|
|
|
// 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->req = r->copy()->as_http();
|
|
|
|
streamHandlers[sid] = entry;
|
|
|
|
// mount the http flv stream.
|
|
// we must register the handler, then start the thread,
|
|
// for the thread will cause thread switch context.
|
|
if ((err = mux.handle(mount, entry->stream)) != srs_success) {
|
|
return srs_error_wrap(err, "http: mount flv stream for vhost=%s failed", sid.c_str());
|
|
}
|
|
|
|
// start http stream cache thread
|
|
if ((err = entry->cache->start()) != srs_success) {
|
|
return srs_error_wrap(err, "http: start stream cache failed");
|
|
}
|
|
srs_trace("http: mount flv stream for sid=%s, mount=%s", sid.c_str(), mount.c_str());
|
|
} else {
|
|
// The entry exists, we reuse it and update the request of stream and cache.
|
|
entry = streamHandlers[sid];
|
|
entry->stream->update_auth(r);
|
|
entry->cache->update_auth(r);
|
|
}
|
|
|
|
if (entry->stream) {
|
|
entry->stream->entry->enabled = true;
|
|
return err;
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
void SrsHttpStreamServer::http_unmount(SrsRequest* r)
|
|
{
|
|
std::string sid = r->get_stream_url();
|
|
|
|
std::map<std::string, SrsLiveEntry*>::iterator it = streamHandlers.find(sid);
|
|
if (it == streamHandlers.end()) {
|
|
return;
|
|
}
|
|
|
|
// Free all HTTP resources.
|
|
SrsUniquePtr<SrsLiveEntry> entry(it->second);
|
|
streamHandlers.erase(it);
|
|
|
|
SrsUniquePtr<SrsLiveStream> stream(entry->stream);
|
|
SrsUniquePtr<SrsBufferCache> cache(entry->cache);
|
|
|
|
// Notify cache and stream to stop.
|
|
if (stream->entry) stream->entry->enabled = false;
|
|
cache->stop();
|
|
|
|
// Wait for cache and stream to stop.
|
|
int i = 0;
|
|
for (; i < 1024; i++) {
|
|
if (!cache->alive() && !stream->alive()) {
|
|
break;
|
|
}
|
|
srs_usleep(100 * SRS_UTIME_MILLISECONDS);
|
|
}
|
|
|
|
// Unmount the HTTP handler, which will free the entry. Note that we must free it after cache and
|
|
// stream stopped for it uses it.
|
|
mux.unhandle(entry->mount, stream.get());
|
|
|
|
srs_trace("http: unmount flv stream for sid=%s, i=%d", sid.c_str(), i);
|
|
}
|
|
|
|
srs_error_t SrsHttpStreamServer::hijack(ISrsHttpMessage* request, ISrsHttpHandler** ph)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// when handler not the root, we think the handler is ok.
|
|
ISrsHttpHandler* h = *ph? *ph : NULL;
|
|
if (h && h->entry && h->entry->pattern != "/") {
|
|
return err;
|
|
}
|
|
|
|
// only hijack for http streaming, http-flv/ts/mp3/aac.
|
|
std::string ext = request->ext();
|
|
if (ext.empty()) {
|
|
return err;
|
|
}
|
|
|
|
// find the actually request vhost.
|
|
SrsConfDirective* vhost = _srs_config->get_vhost(request->host());
|
|
if (!vhost || !_srs_config->get_vhost_enabled(vhost)) {
|
|
return err;
|
|
}
|
|
|
|
// find the entry template for the stream.
|
|
SrsLiveEntry* entry = NULL;
|
|
if (true) {
|
|
// no http streaming on vhost, ignore.
|
|
std::map<std::string, SrsLiveEntry*>::iterator it = templateHandlers.find(vhost->arg0());
|
|
if (it == templateHandlers.end()) {
|
|
return err;
|
|
}
|
|
|
|
// hstrs always enabled.
|
|
// for origin, the http stream will be mount already when publish,
|
|
// so it must never enter this line for stream already mounted.
|
|
// for edge, the http stream is trigger by hstrs and mount by it,
|
|
// so we only hijack when only edge and hstrs is on.
|
|
entry = it->second;
|
|
|
|
// check entry and request extension.
|
|
if (entry->is_flv()) {
|
|
if (ext != ".flv") {
|
|
return err;
|
|
}
|
|
} else if (entry->is_ts()) {
|
|
if (ext != ".ts") {
|
|
return err;
|
|
}
|
|
} else if (entry->is_mp3()) {
|
|
if (ext != ".mp3") {
|
|
return err;
|
|
}
|
|
} else if (entry->is_aac()) {
|
|
if (ext != ".aac") {
|
|
return err;
|
|
}
|
|
} else {
|
|
return err;
|
|
}
|
|
}
|
|
|
|
// For HTTP-FLV stream, the template must have the same schema with upath.
|
|
// The template is defined in config, the mout of http stream. The upath is specified by http request path.
|
|
// If template is "[vhost]/[app]/[stream].flv", the upath should be:
|
|
// matched for "/live/livestream.flv"
|
|
// matched for "ossrs.net/live/livestream.flv"
|
|
// not-matched for "/livestream.flv", which is actually "/__defaultApp__/livestream.flv", HTTP not support default app.
|
|
// not-matched for "/live/show/livestream.flv"
|
|
string upath = request->path();
|
|
if (srs_string_count(upath, "/") != srs_string_count(entry->mount, "/")) {
|
|
return err;
|
|
}
|
|
|
|
// convert to concreate class.
|
|
SrsHttpMessage* hreq = dynamic_cast<SrsHttpMessage*>(request);
|
|
srs_assert(hreq);
|
|
|
|
// hijack for entry.
|
|
SrsUniquePtr<SrsRequest> r(hreq->to_request(vhost->arg0()));
|
|
|
|
std::string sid = r->get_stream_url();
|
|
// check whether the http remux is enabled,
|
|
// for example, user disable the http flv then reload.
|
|
if (streamHandlers.find(sid) != streamHandlers.end()) {
|
|
SrsLiveEntry* s_entry = streamHandlers[sid];
|
|
if (!s_entry->stream->entry->enabled) {
|
|
// only when the http entry is disabled, check the config whether http flv disable,
|
|
// for the http flv edge use hijack to trigger the edge ingester, we always mount it
|
|
// eventhough the origin does not exists the specified stream.
|
|
if (!_srs_config->get_vhost_http_remux_enabled(r->vhost)) {
|
|
return srs_error_new(ERROR_HTTP_HIJACK, "stream disabled");
|
|
}
|
|
}
|
|
}
|
|
|
|
SrsSharedPtr<SrsLiveSource> live_source;
|
|
if ((err = _srs_sources->fetch_or_create(r.get(), server, live_source)) != srs_success) {
|
|
return srs_error_wrap(err, "source create");
|
|
}
|
|
srs_assert(live_source.get() != NULL);
|
|
|
|
bool enabled_cache = _srs_config->get_gop_cache(r->vhost);
|
|
int gcmf = _srs_config->get_gop_cache_max_frames(r->vhost);
|
|
live_source->set_cache(enabled_cache);
|
|
live_source->set_gop_cache_max_frames(gcmf);
|
|
|
|
// create http streaming handler.
|
|
if ((err = http_mount(r.get())) != srs_success) {
|
|
return srs_error_wrap(err, "http mount");
|
|
}
|
|
|
|
// use the handler if exists.
|
|
if (streamHandlers.find(sid) != streamHandlers.end()) {
|
|
entry = streamHandlers[sid];
|
|
*ph = entry->stream;
|
|
}
|
|
|
|
// trigger edge to fetch from origin.
|
|
bool vhost_is_edge = _srs_config->get_vhost_is_edge(r->vhost);
|
|
srs_trace("flv: source url=%s, is_edge=%d, source_id=%s/%s",
|
|
r->get_stream_url().c_str(), vhost_is_edge, live_source->source_id().c_str(), live_source->pre_source_id().c_str());
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsHttpStreamServer::initialize_flv_streaming()
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// http flv live stream mount for each vhost.
|
|
SrsConfDirective* root = _srs_config->get_root();
|
|
for (int i = 0; i < (int)root->directives.size(); i++) {
|
|
SrsConfDirective* conf = root->at(i);
|
|
|
|
if (!conf->is_vhost()) {
|
|
continue;
|
|
}
|
|
|
|
if ((err = initialize_flv_entry(conf->arg0())) != srs_success) {
|
|
return srs_error_wrap(err, "init flv entries");
|
|
}
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsHttpStreamServer::initialize_flv_entry(std::string vhost)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
if (!_srs_config->get_vhost_http_remux_enabled(vhost)) {
|
|
return err;
|
|
}
|
|
|
|
SrsLiveEntry* entry = new SrsLiveEntry(_srs_config->get_vhost_http_remux_mount(vhost));
|
|
|
|
templateHandlers[vhost] = entry;
|
|
srs_trace("http flv live stream, vhost=%s, mount=%s", vhost.c_str(), entry->mount.c_str());
|
|
|
|
return err;
|
|
}
|
|
|