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srs/trunk/src/app/srs_app_http_stream.cpp
Haibo Chen 58e775ce8d
HLS: Fix error when stream has extension. #4215 v5.0.217 v6.0.158 v7.0.19 (#4216)
---------

Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
2024-10-31 17:50:56 +08:00

1350 lines
39 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>
#include <srs_app_async_call.hpp>
SrsBufferCache::SrsBufferCache(SrsServer* s, 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);
server_ = s;
}
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;
// Not enabled.
if (fast_cache <= 0) {
return err;
}
if ((err = trd->start()) != srs_success) {
return srs_error_wrap(err, "corotine");
}
return err;
}
void SrsBufferCache::stop()
{
// Not enabled.
if (fast_cache <= 0) {
return;
}
trd->stop();
}
bool SrsBufferCache::alive()
{
// Not enabled.
if (fast_cache <= 0) {
return false;
}
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;
SrsSharedPtr<SrsLiveSource> live_source;
if ((err = _srs_sources->fetch_or_create(req, server_, live_source)) != srs_success) {
return srs_error_wrap(err, "source create");
}
srs_assert(live_source.get() != NULL);
// 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(SrsServer* s, SrsRequest* r, SrsBufferCache* c)
{
cache = c;
req = r->copy()->as_http();
security_ = new SrsSecurity();
server_ = s;
}
SrsLiveStream::~SrsLiveStream()
{
srs_freep(req);
srs_freep(security_);
// The live stream should never be destroyed when it's serving any viewers.
srs_assert(viewers_.empty());
}
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());
// remove the extension of stream if have. for instance, test.flv -> test
req->stream = srs_path_filename(req->stream);
// 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");
}
// Always try to create the source, because http handler won't create it.
SrsSharedPtr<SrsLiveSource> live_source;
if ((err = _srs_sources->fetch_or_create(req, 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(req->vhost);
int gcmf = _srs_config->get_gop_cache_max_frames(req->vhost);
live_source->set_cache(enabled_cache);
live_source->set_gop_cache_max_frames(gcmf);
// Create consumer of source, 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");
}
// When freeing the consumer, it may trigger the source unpublishing for edge. This will trigger the http
// unmount, which waiting for all http live stream to dispose, so we should free the consumer when this
// object is not alive.
SrsUniquePtr<SrsLiveConsumer> consumer(consumer_raw);
// Add the viewer to the viewers list.
viewers_.push_back(hc);
// Serve the viewer connection.
err = do_serve_http(live_source.get(), consumer.get(), w, r);
// Remove viewer from the viewers list.
vector<ISrsExpire*>::iterator it = std::find(viewers_.begin(), viewers_.end(), hc);
srs_assert (it != viewers_.end());
viewers_.erase(it);
// Do hook after serving.
http_hooks_on_stop(r);
return err;
}
bool SrsLiveStream::alive()
{
return !viewers_.empty();
}
void SrsLiveStream::expire()
{
vector<ISrsExpire*>::iterator it;
for (it = viewers_.begin(); it != viewers_.end(); ++it) {
ISrsExpire* conn = *it;
conn->expire();
}
}
srs_error_t SrsLiveStream::do_serve_http(SrsLiveSource* source, SrsLiveConsumer* consumer, 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);
if ((err = source->consumer_dumps(consumer, 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, 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;
disposing = false;
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;
async_ = new SrsAsyncCallWorker();
mux.hijack(this);
_srs_config->subscribe(this);
}
SrsHttpStreamServer::~SrsHttpStreamServer()
{
mux.unhijack(this);
_srs_config->unsubscribe(this);
async_->stop();
srs_freep(async_);
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");
}
if ((err = async_->start()) != srs_success) {
return srs_error_wrap(err, "async start");
}
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(server, r);
entry->stream = new SrsLiveStream(server, 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];
// Fail if system is disposing the entry.
if (entry->disposing) {
return srs_error_new(ERROR_STREAM_DISPOSING, "stream is disposing");
}
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;
}
// Set the entry to disposing, which will prevent the stream to be reused.
SrsLiveEntry* entry = it->second;
if (entry->disposing) {
return;
}
entry->disposing = true;
// Use async worker to execute the task, which will destroy the stream.
srs_error_t err = srs_success;
if ((err = async_->execute(new SrsHttpStreamDestroy(&mux, &streamHandlers, sid))) != srs_success) {
srs_warn("http: ignore unmount stream failed, sid=%s, err=%s", sid.c_str(), srs_error_desc(err).c_str());
srs_freep(err);
}
}
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");
}
}
}
// 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;
}
srs_trace("flv: hijack %s ok", upath.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;
}
SrsHttpStreamDestroy::SrsHttpStreamDestroy(SrsHttpServeMux* mux, map<std::string, SrsLiveEntry*>* handlers, string sid)
{
mux_ = mux;
sid_ = sid;
streamHandlers_ = handlers;
}
SrsHttpStreamDestroy::~SrsHttpStreamDestroy()
{
}
srs_error_t SrsHttpStreamDestroy::call()
{
srs_error_t err = srs_success;
std::map<std::string, SrsLiveEntry*>::iterator it = streamHandlers_->find(sid_);
if (it == streamHandlers_->end()) {
return err;
}
// Free all HTTP resources.
SrsUniquePtr<SrsLiveEntry> entry(it->second);
srs_assert(entry->disposing);
SrsUniquePtr<SrsLiveStream> stream(entry->stream);
SrsUniquePtr<SrsBufferCache> cache(entry->cache);
// Notify cache and stream to stop.
if (stream->entry) stream->entry->enabled = false;
stream->expire();
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);
}
if (cache->alive() || stream->alive()) {
srs_warn("http: try to free a alive stream, cache=%d, stream=%d", cache->alive(), stream->alive());
}
// Remove the entry from handlers.
streamHandlers_->erase(it);
// 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);
return err;
}
string SrsHttpStreamDestroy::to_string()
{
return "destroy";
}