/** * The MIT License (MIT) * * Copyright (c) 2013-2020 Winlin * * Permission is hereby granted, free of charge, to any person obtaining a copy of * this software and associated documentation files (the "Software"), to deal in * the Software without restriction, including without limitation the rights to * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of * the Software, and to permit persons to whom the Software is furnished to do so, * subject to the following conditions: * * The above copyright notice and this permission notice shall be included in all * copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS * FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR * COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER * IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */ #include #include #include #include #include #include #include #include #include #include using namespace std; #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include // drop the segment when duration of ts too small. // TODO: FIXME: Refine to time unit. #define SRS_AUTO_HLS_SEGMENT_MIN_DURATION (100 * SRS_UTIME_MILLISECONDS) // fragment plus the deviation percent. #define SRS_HLS_FLOOR_REAP_PERCENT 0.3 // reset the piece id when deviation overflow this. #define SRS_JUMP_WHEN_PIECE_DEVIATION 20 SrsHlsSegment::SrsHlsSegment(SrsTsContext* c, SrsAudioCodecId ac, SrsVideoCodecId vc, SrsFileWriter* w) { sequence_no = 0; writer = w; tscw = new SrsTsContextWriter(writer, c, ac, vc); } SrsHlsSegment::~SrsHlsSegment() { srs_freep(tscw); } void SrsHlsSegment::config_cipher(unsigned char* key,unsigned char* iv) { memcpy(this->iv, iv,16); SrsEncFileWriter* fw = (SrsEncFileWriter*)writer; fw->config_cipher(key, iv); } SrsDvrAsyncCallOnHls::SrsDvrAsyncCallOnHls(int c, SrsRequest* r, string p, string t, string m, string mu, int s, srs_utime_t d) { req = r->copy(); cid = c; path = p; ts_url = t; m3u8 = m; m3u8_url = mu; seq_no = s; duration = d; } SrsDvrAsyncCallOnHls::~SrsDvrAsyncCallOnHls() { srs_freep(req); } srs_error_t SrsDvrAsyncCallOnHls::call() { srs_error_t err = srs_success; if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost)) { return err; } // 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 hooks; if (true) { SrsConfDirective* conf = _srs_config->get_vhost_on_hls(req->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_hls(cid, url, req, path, ts_url, m3u8, m3u8_url, seq_no, duration)) != srs_success) { return srs_error_wrap(err, "callback on_hls %s", url.c_str()); } } return err; } string SrsDvrAsyncCallOnHls::to_string() { return "on_hls: " + path; } SrsDvrAsyncCallOnHlsNotify::SrsDvrAsyncCallOnHlsNotify(int c, SrsRequest* r, string u) { cid = c; req = r->copy(); ts_url = u; } SrsDvrAsyncCallOnHlsNotify::~SrsDvrAsyncCallOnHlsNotify() { srs_freep(req); } srs_error_t SrsDvrAsyncCallOnHlsNotify::call() { srs_error_t err = srs_success; if (!_srs_config->get_vhost_http_hooks_enabled(req->vhost)) { return err; } // 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 hooks; if (true) { SrsConfDirective* conf = _srs_config->get_vhost_on_hls_notify(req->vhost); if (!conf) { return err; } hooks = conf->args; } int nb_notify = _srs_config->get_vhost_hls_nb_notify(req->vhost); for (int i = 0; i < (int)hooks.size(); i++) { std::string url = hooks.at(i); if ((err = SrsHttpHooks::on_hls_notify(cid, url, req, ts_url, nb_notify)) != srs_success) { return srs_error_wrap(err, "callback on_hls_notify %s", url.c_str()); } } return err; } string SrsDvrAsyncCallOnHlsNotify::to_string() { return "on_hls_notify: " + ts_url; } SrsHlsMuxer::SrsHlsMuxer() { req = NULL; hls_fragment = hls_window = 0; hls_aof_ratio = 1.0; deviation_ts = 0; hls_cleanup = true; hls_wait_keyframe = true; previous_floor_ts = 0; accept_floor_ts = 0; hls_ts_floor = false; max_td = 0; _sequence_no = 0; current = NULL; hls_keys = false; hls_fragments_per_key = 0; async = new SrsAsyncCallWorker(); context = new SrsTsContext(); segments = new SrsFragmentWindow(); memset(key, 0, 16); memset(iv, 0, 16); } SrsHlsMuxer::~SrsHlsMuxer() { srs_freep(current); srs_freep(req); srs_freep(async); srs_freep(context); srs_freep(writer); } void SrsHlsMuxer::dispose() { srs_error_t err = srs_success; segments->dispose(); if (current) { if ((err = current->unlink_tmpfile()) != srs_success) { srs_warn("Unlink tmp ts failed %s", srs_error_desc(err).c_str()); srs_freep(err); } srs_freep(current); } if (unlink(m3u8.c_str()) < 0) { srs_warn("dispose unlink path failed. file=%s", m3u8.c_str()); } srs_trace("gracefully dispose hls %s", req? req->get_stream_url().c_str() : ""); } int SrsHlsMuxer::sequence_no() { return _sequence_no; } string SrsHlsMuxer::ts_url() { return current? current->uri:""; } srs_utime_t SrsHlsMuxer::duration() { return current? current->duration():0; } int SrsHlsMuxer::deviation() { // no floor, no deviation. if (!hls_ts_floor) { return 0; } return deviation_ts; } srs_error_t SrsHlsMuxer::initialize() { return srs_success; } srs_error_t SrsHlsMuxer::on_publish(SrsRequest* req) { srs_error_t err = srs_success; if ((err = async->start()) != srs_success) { return srs_error_wrap(err, "async start"); } return err; } srs_error_t SrsHlsMuxer::on_unpublish() { async->stop(); return srs_success; } srs_error_t SrsHlsMuxer::update_config(SrsRequest* r, string entry_prefix, string path, string m3u8_file, string ts_file, srs_utime_t fragment, srs_utime_t window, bool ts_floor, double aof_ratio, bool cleanup, bool wait_keyframe, bool keys, int fragments_per_key, string key_file ,string key_file_path, string key_url) { srs_error_t err = srs_success; srs_freep(req); req = r->copy(); hls_entry_prefix = entry_prefix; hls_path = path; hls_ts_file = ts_file; hls_fragment = fragment; hls_aof_ratio = aof_ratio; hls_ts_floor = ts_floor; hls_cleanup = cleanup; hls_wait_keyframe = wait_keyframe; previous_floor_ts = 0; accept_floor_ts = 0; hls_window = window; deviation_ts = 0; hls_keys = keys; hls_fragments_per_key = fragments_per_key; hls_key_file = key_file; hls_key_file_path = key_file_path; hls_key_url = key_url; // generate the m3u8 dir and path. m3u8_url = srs_path_build_stream(m3u8_file, req->vhost, req->app, req->stream); m3u8 = path + "/" + m3u8_url; // when update config, reset the history target duration. max_td = fragment * _srs_config->get_hls_td_ratio(r->vhost); // create m3u8 dir once. m3u8_dir = srs_path_dirname(m3u8); if ((err = srs_create_dir_recursively(m3u8_dir)) != srs_success) { return srs_error_wrap(err, "create dir"); } if (hls_keys && (hls_path != hls_key_file_path)) { string key_file = srs_path_build_stream(hls_key_file, req->vhost, req->app, req->stream); string key_url = hls_key_file_path + "/" + key_file; string key_dir = srs_path_dirname(key_url); if ((err = srs_create_dir_recursively(key_dir)) != srs_success) { return srs_error_wrap(err, "create dir"); } } if(hls_keys) { writer = new SrsEncFileWriter(); } else { writer = new SrsFileWriter(); } return err; } srs_error_t SrsHlsMuxer::segment_open() { srs_error_t err = srs_success; if (current) { srs_warn("ignore the segment open, for segment is already open."); return err; } // when segment open, the current segment must be NULL. srs_assert(!current); // load the default acodec from config. SrsAudioCodecId default_acodec = SrsAudioCodecIdAAC; if (true) { std::string default_acodec_str = _srs_config->get_hls_acodec(req->vhost); if (default_acodec_str == "mp3") { default_acodec = SrsAudioCodecIdMP3; } else if (default_acodec_str == "aac") { default_acodec = SrsAudioCodecIdAAC; } else if (default_acodec_str == "an") { default_acodec = SrsAudioCodecIdDisabled; } else { srs_warn("hls: use aac for other codec=%s", default_acodec_str.c_str()); } } // load the default vcodec from config. SrsVideoCodecId default_vcodec = SrsVideoCodecIdAVC; if (true) { std::string default_vcodec_str = _srs_config->get_hls_vcodec(req->vhost); if (default_vcodec_str == "h264") { default_vcodec = SrsVideoCodecIdAVC; } else if (default_vcodec_str == "vn") { default_vcodec = SrsVideoCodecIdDisabled; } else { srs_warn("hls: use h264 for other codec=%s", default_vcodec_str.c_str()); } } // new segment. current = new SrsHlsSegment(context, default_acodec, default_vcodec, writer); current->sequence_no = _sequence_no++; if ((err = write_hls_key()) != srs_success) { return srs_error_wrap(err, "write hls key"); } // generate filename. std::string ts_file = hls_ts_file; ts_file = srs_path_build_stream(ts_file, req->vhost, req->app, req->stream); if (hls_ts_floor) { // accept the floor ts for the first piece. int64_t current_floor_ts = srs_update_system_time() / hls_fragment; if (!accept_floor_ts) { accept_floor_ts = current_floor_ts - 1; } else { accept_floor_ts++; } // jump when deviation more than 10p if (accept_floor_ts - current_floor_ts > SRS_JUMP_WHEN_PIECE_DEVIATION) { srs_warn("hls: jmp for ts deviation, current=%" PRId64 ", accept=%" PRId64, current_floor_ts, accept_floor_ts); accept_floor_ts = current_floor_ts - 1; } // when reap ts, adjust the deviation. deviation_ts = (int)(accept_floor_ts - current_floor_ts); // dup/jmp detect for ts in floor mode. if (previous_floor_ts && previous_floor_ts != current_floor_ts - 1) { srs_warn("hls: dup/jmp ts, previous=%" PRId64 ", current=%" PRId64 ", accept=%" PRId64 ", deviation=%d", previous_floor_ts, current_floor_ts, accept_floor_ts, deviation_ts); } previous_floor_ts = current_floor_ts; // we always ensure the piece is increase one by one. std::stringstream ts_floor; ts_floor << accept_floor_ts; ts_file = srs_string_replace(ts_file, "[timestamp]", ts_floor.str()); // TODO: FIMXE: we must use the accept ts floor time to generate the hour variable. ts_file = srs_path_build_timestamp(ts_file); } else { ts_file = srs_path_build_timestamp(ts_file); } if (true) { std::stringstream ss; ss << current->sequence_no; ts_file = srs_string_replace(ts_file, "[seq]", ss.str()); } current->set_path(hls_path + "/" + ts_file); // the ts url, relative or absolute url. // TODO: FIXME: Use url and path manager. std::string ts_url = current->fullpath(); if (srs_string_starts_with(ts_url, m3u8_dir)) { ts_url = ts_url.substr(m3u8_dir.length()); } while (srs_string_starts_with(ts_url, "/")) { ts_url = ts_url.substr(1); } current->uri += hls_entry_prefix; if (!hls_entry_prefix.empty() && !srs_string_ends_with(hls_entry_prefix, "/")) { current->uri += "/"; // add the http dir to uri. string http_dir = srs_path_dirname(m3u8_url); if (!http_dir.empty()) { current->uri += http_dir + "/"; } } current->uri += ts_url; // create dir recursively for hls. if ((err = current->create_dir()) != srs_success) { return srs_error_wrap(err, "create dir"); } // open temp ts file. std::string tmp_file = current->tmppath(); if ((err = current->writer->open(tmp_file)) != srs_success) { return srs_error_wrap(err, "open hls muxer"); } // reset the context for a new ts start. context->reset(); return err; } srs_error_t SrsHlsMuxer::on_sequence_header() { srs_error_t err = srs_success; srs_assert(current); // set the current segment to sequence header, // when close the segement, it will write a discontinuity to m3u8 file. current->set_sequence_header(true); return err; } bool SrsHlsMuxer::is_segment_overflow() { srs_assert(current); // to prevent very small segment. if (current->duration() < 2 * SRS_AUTO_HLS_SEGMENT_MIN_DURATION) { return false; } // use N% deviation, to smoother. srs_utime_t deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0; return current->duration() >= hls_fragment + deviation; } bool SrsHlsMuxer::wait_keyframe() { return hls_wait_keyframe; } bool SrsHlsMuxer::is_segment_absolutely_overflow() { // @see https://github.com/ossrs/srs/issues/151#issuecomment-83553950 srs_assert(current); // to prevent very small segment. if (current->duration() < 2 * SRS_AUTO_HLS_SEGMENT_MIN_DURATION) { return false; } // use N% deviation, to smoother. srs_utime_t deviation = hls_ts_floor? SRS_HLS_FLOOR_REAP_PERCENT * deviation_ts * hls_fragment : 0; return current->duration() >= hls_aof_ratio * hls_fragment + deviation; } bool SrsHlsMuxer::pure_audio() { return current && current->tscw && current->tscw->video_codec() == SrsVideoCodecIdDisabled; } srs_error_t SrsHlsMuxer::flush_audio(SrsTsMessageCache* cache) { srs_error_t err = srs_success; // if current is NULL, segment is not open, ignore the flush event. if (!current) { srs_warn("flush audio ignored, for segment is not open."); return err; } if (!cache->audio || cache->audio->payload->length() <= 0) { return err; } // update the duration of segment. current->append(cache->audio->pts / 90); if ((err = current->tscw->write_audio(cache->audio)) != srs_success) { return srs_error_wrap(err, "hls: write audio"); } // write success, clear and free the msg srs_freep(cache->audio); return err; } srs_error_t SrsHlsMuxer::flush_video(SrsTsMessageCache* cache) { srs_error_t err = srs_success; // if current is NULL, segment is not open, ignore the flush event. if (!current) { srs_warn("flush video ignored, for segment is not open."); return err; } if (!cache->video || cache->video->payload->length() <= 0) { return err; } srs_assert(current); // update the duration of segment. current->append(cache->video->dts / 90); if ((err = current->tscw->write_video(cache->video)) != srs_success) { return srs_error_wrap(err, "hls: write video"); } // write success, clear and free the msg srs_freep(cache->video); return err; } srs_error_t SrsHlsMuxer::segment_close() { srs_error_t err = do_segment_close(); // We always cleanup current segment. srs_freep(current); return err; } srs_error_t SrsHlsMuxer::do_segment_close() { srs_error_t err = srs_success; if (!current) { srs_warn("ignore the segment close, for segment is not open."); return err; } // when close current segment, the current segment must not be NULL. srs_assert(current); // We should always close the underlayer writer. if (current && current->writer) { current->writer->close(); } // valid, add to segments if segment duration is ok // when too small, it maybe not enough data to play. // when too large, it maybe timestamp corrupt. // make the segment more acceptable, when in [min, max_td * 2], it's ok. bool matchMinDuration = current->duration() >= SRS_AUTO_HLS_SEGMENT_MIN_DURATION; bool matchMaxDuration = current->duration() <= max_td * 2 * 1000; if (matchMinDuration && matchMaxDuration) { // use async to call the http hooks, for it will cause thread switch. if ((err = async->execute(new SrsDvrAsyncCallOnHls(_srs_context->get_id(), req, current->fullpath(), current->uri, m3u8, m3u8_url, current->sequence_no, current->duration()))) != srs_success) { return srs_error_wrap(err, "segment close"); } // use async to call the http hooks, for it will cause thread switch. if ((err = async->execute(new SrsDvrAsyncCallOnHlsNotify(_srs_context->get_id(), req, current->uri))) != srs_success) { return srs_error_wrap(err, "segment close"); } // close the muxer of finished segment. srs_freep(current->tscw); // rename from tmp to real path if ((err = current->rename()) != srs_success) { return srs_error_wrap(err, "rename"); } segments->append(current); current = NULL; } else { // reuse current segment index. _sequence_no--; srs_trace("Drop ts segment, sequence_no=%d, uri=%s, duration=%dms", current->sequence_no, current->uri.c_str(), srsu2msi(current->duration())); // rename from tmp to real path if ((err = current->unlink_tmpfile()) != srs_success) { return srs_error_wrap(err, "rename"); } } // shrink the segments. segments->shrink(hls_window); // refresh the m3u8, donot contains the removed ts err = refresh_m3u8(); // remove the ts file. segments->clear_expired(hls_cleanup); // check ret of refresh m3u8 if (err != srs_success) { return srs_error_wrap(err, "hls: refresh m3u8"); } return err; } srs_error_t SrsHlsMuxer::write_hls_key() { srs_error_t err = srs_success; if (hls_keys && current->sequence_no % hls_fragments_per_key == 0) { if (RAND_bytes(key, 16) < 0) { return srs_error_wrap(err, "rand key failed."); } if (RAND_bytes(iv, 16) < 0) { return srs_error_wrap(err, "rand iv failed."); } string key_file = srs_path_build_stream(hls_key_file, req->vhost, req->app, req->stream); key_file = srs_string_replace(key_file, "[seq]", srs_int2str(current->sequence_no)); string key_url = hls_key_file_path + "/" + key_file; SrsFileWriter fw; if ((err = fw.open(key_url)) != srs_success) { return srs_error_wrap(err, "open file %s", key_url.c_str()); } err = fw.write(key, 16, NULL); fw.close(); if (err != srs_success) { return srs_error_wrap(err, "write key"); } } if (hls_keys) { current->config_cipher(key, iv); } return err; } srs_error_t SrsHlsMuxer::refresh_m3u8() { srs_error_t err = srs_success; // no segments, also no m3u8, return. if (segments->empty()) { return err; } std::string temp_m3u8 = m3u8 + ".temp"; if ((err = _refresh_m3u8(temp_m3u8)) == srs_success) { if (rename(temp_m3u8.c_str(), m3u8.c_str()) < 0) { err = srs_error_new(ERROR_HLS_WRITE_FAILED, "hls: rename m3u8 file failed. %s => %s", temp_m3u8.c_str(), m3u8.c_str()); } } // remove the temp file. 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 err; } srs_error_t SrsHlsMuxer::_refresh_m3u8(string m3u8_file) { srs_error_t err = srs_success; // no segments, return. if (segments->empty()) { return err; } SrsFileWriter writer; if ((err = writer.open(m3u8_file)) != srs_success) { return srs_error_wrap(err, "hls: open m3u8 file %s", m3u8_file.c_str()); } // #EXTM3U\n // #EXT-X-VERSION:3\n std::stringstream ss; ss << "#EXTM3U" << SRS_CONSTS_LF; ss << "#EXT-X-VERSION:3" << SRS_CONSTS_LF; // #EXT-X-MEDIA-SEQUENCE:4294967295\n SrsHlsSegment* first = dynamic_cast(segments->first()); ss << "#EXT-X-MEDIA-SEQUENCE:" << first->sequence_no << SRS_CONSTS_LF; // iterator shared for td generation and segemnts wrote. std::vector::iterator it; // #EXT-X-TARGETDURATION:4294967295\n /** * @see hls-m3u8-draft-pantos-http-live-streaming-12.pdf, page 25 * The Media Playlist file MUST contain an EXT-X-TARGETDURATION tag. * Its value MUST be equal to or greater than the EXTINF duration of any * media segment that appears or will appear in the Playlist file, * rounded to the nearest integer. Its value MUST NOT change. A * typical target duration is 10 seconds. */ // @see https://github.com/ossrs/srs/issues/304#issuecomment-74000081 srs_utime_t max_duration = segments->max_duration(); int target_duration = (int)ceil(srsu2msi(srs_max(max_duration, max_td)) / 1000.0); ss << "#EXT-X-TARGETDURATION:" << target_duration << SRS_CONSTS_LF; // write all segments for (int i = 0; i < segments->size(); i++) { SrsHlsSegment* segment = dynamic_cast(segments->at(i)); if (segment->is_sequence_header()) { // #EXT-X-DISCONTINUITY\n ss << "#EXT-X-DISCONTINUITY" << SRS_CONSTS_LF; } if(hls_keys && ((segment->sequence_no % hls_fragments_per_key) == 0)) { char hexiv[33]; srs_data_to_hex(hexiv, segment->iv, 16); hexiv[32] = '\0'; string key_file = srs_path_build_stream(hls_key_file, req->vhost, req->app, req->stream); key_file = srs_string_replace(key_file, "[seq]", srs_int2str(segment->sequence_no)); string key_path = key_file; //if key_url is not set,only use the file name if (!hls_key_url.empty()) { key_path = hls_key_url + key_file; } ss << "#EXT-X-KEY:METHOD=AES-128,URI=" << "\"" << key_path << "\",IV=0x" << hexiv << SRS_CONSTS_LF; } // "#EXTINF:4294967295.208,\n" ss.precision(3); ss.setf(std::ios::fixed, std::ios::floatfield); ss << "#EXTINF:" << srsu2msi(segment->duration()) / 1000.0 << ", no desc" << SRS_CONSTS_LF; // {file name}\n std::string seg_uri = segment->uri; if (true) { std::stringstream stemp; stemp << srsu2msi(segment->duration()); seg_uri = srs_string_replace(seg_uri, "[duration]", stemp.str()); } //ss << segment->uri << SRS_CONSTS_LF; ss << seg_uri << SRS_CONSTS_LF; } // write m3u8 to writer. std::string m3u8 = ss.str(); if ((err = writer.write((char*)m3u8.c_str(), (int)m3u8.length(), NULL)) != srs_success) { return srs_error_wrap(err, "hls: write m3u8"); } return err; } SrsHlsController::SrsHlsController() { tsmc = new SrsTsMessageCache(); muxer = new SrsHlsMuxer(); } SrsHlsController::~SrsHlsController() { srs_freep(muxer); srs_freep(tsmc); } srs_error_t SrsHlsController::initialize() { srs_error_t err = muxer->initialize(); if (err != srs_success) { return srs_error_wrap(err, "hls muxer initialize"); } return srs_success; } void SrsHlsController::dispose() { muxer->dispose(); } int SrsHlsController::sequence_no() { return muxer->sequence_no(); } string SrsHlsController::ts_url() { return muxer->ts_url(); } srs_utime_t SrsHlsController::duration() { return muxer->duration(); } int SrsHlsController::deviation() { return muxer->deviation(); } srs_error_t SrsHlsController::on_publish(SrsRequest* req) { srs_error_t err = srs_success; std::string vhost = req->vhost; std::string stream = req->stream; std::string app = req->app; srs_utime_t hls_fragment = _srs_config->get_hls_fragment(vhost); srs_utime_t hls_window = _srs_config->get_hls_window(vhost); // get the hls m3u8 ts list entry prefix config std::string entry_prefix = _srs_config->get_hls_entry_prefix(vhost); // get the hls path config std::string path = _srs_config->get_hls_path(vhost); std::string m3u8_file = _srs_config->get_hls_m3u8_file(vhost); std::string ts_file = _srs_config->get_hls_ts_file(vhost); bool cleanup = _srs_config->get_hls_cleanup(vhost); bool wait_keyframe = _srs_config->get_hls_wait_keyframe(vhost); // the audio overflow, for pure audio to reap segment. double hls_aof_ratio = _srs_config->get_hls_aof_ratio(vhost); // whether use floor(timestamp/hls_fragment) for variable timestamp bool ts_floor = _srs_config->get_hls_ts_floor(vhost); // the seconds to dispose the hls. srs_utime_t hls_dispose = _srs_config->get_hls_dispose(vhost); bool hls_keys = _srs_config->get_hls_keys(vhost); int hls_fragments_per_key = _srs_config->get_hls_fragments_per_key(vhost); string hls_key_file = _srs_config->get_hls_key_file(vhost); string hls_key_file_path = _srs_config->get_hls_key_file_path(vhost); string hls_key_url = _srs_config->get_hls_key_url(vhost); // TODO: FIXME: support load exists m3u8, to continue publish stream. // for the HLS donot requires the EXT-X-MEDIA-SEQUENCE be monotonically increase. if ((err = muxer->on_publish(req)) != srs_success) { return srs_error_wrap(err, "muxer publish"); } if ((err = muxer->update_config(req, entry_prefix, path, m3u8_file, ts_file, hls_fragment, hls_window, ts_floor, hls_aof_ratio, cleanup, wait_keyframe,hls_keys,hls_fragments_per_key, hls_key_file, hls_key_file_path, hls_key_url)) != srs_success ) { return srs_error_wrap(err, "hls: update config"); } if ((err = muxer->segment_open()) != srs_success) { return srs_error_wrap(err, "hls: segment open"); } // This config item is used in SrsHls, we just log its value here. bool hls_dts_directly = _srs_config->get_vhost_hls_dts_directly(req->vhost); srs_trace("hls: win=%dms, frag=%dms, prefix=%s, path=%s, m3u8=%s, ts=%s, aof=%.2f, floor=%d, clean=%d, waitk=%d, dispose=%dms, dts_directly=%d", srsu2msi(hls_window), srsu2msi(hls_fragment), entry_prefix.c_str(), path.c_str(), m3u8_file.c_str(), ts_file.c_str(), hls_aof_ratio, ts_floor, cleanup, wait_keyframe, srsu2msi(hls_dispose), hls_dts_directly); return err; } srs_error_t SrsHlsController::on_unpublish() { srs_error_t err = srs_success; if ((err = muxer->on_unpublish()) != srs_success) { return srs_error_wrap(err, "muxer unpublish"); } if ((err = muxer->flush_audio(tsmc)) != srs_success) { return srs_error_wrap(err, "hls: flush audio"); } if ((err = muxer->segment_close()) != srs_success) { return srs_error_wrap(err, "hls: segment close"); } return err; } srs_error_t SrsHlsController::on_sequence_header() { // TODO: support discontinuity for the same stream // currently we reap and insert discontinity when encoder republish, // but actually, event when stream is not republish, the // sequence header may change, for example, // ffmpeg ingest a external rtmp stream and push to srs, // when the sequence header changed, the stream is not republish. return muxer->on_sequence_header(); } srs_error_t SrsHlsController::write_audio(SrsAudioFrame* frame, int64_t pts) { srs_error_t err = srs_success; // write audio to cache. if ((err = tsmc->cache_audio(frame, pts)) != srs_success) { return srs_error_wrap(err, "hls: cache audio"); } // reap when current source is pure audio. // it maybe changed when stream info changed, // for example, pure audio when start, audio/video when publishing, // pure audio again for audio disabled. // so we reap event when the audio incoming when segment overflow. // @see https://github.com/ossrs/srs/issues/151 // we use absolutely overflow of segment to make jwplayer/ffplay happy // @see https://github.com/ossrs/srs/issues/151#issuecomment-71155184 if (tsmc->audio && muxer->is_segment_absolutely_overflow()) { if ((err = reap_segment()) != srs_success) { return srs_error_wrap(err, "hls: reap segment"); } } // for pure audio, aggregate some frame to one. // TODO: FIXME: Check whether it's necessary. if (muxer->pure_audio() && tsmc->audio) { if (pts - tsmc->audio->start_pts < SRS_CONSTS_HLS_PURE_AUDIO_AGGREGATE) { return err; } } // directly write the audio frame by frame to ts, // it's ok for the hls overload, or maybe cause the audio corrupt, // which introduced by aggregate the audios to a big one. // @see https://github.com/ossrs/srs/issues/512 if ((err = muxer->flush_audio(tsmc)) != srs_success) { return srs_error_wrap(err, "hls: flush audio"); } return err; } srs_error_t SrsHlsController::write_video(SrsVideoFrame* frame, int64_t dts) { srs_error_t err = srs_success; // write video to cache. if ((err = tsmc->cache_video(frame, dts)) != srs_success) { return srs_error_wrap(err, "hls: cache video"); } // when segment overflow, reap if possible. if (muxer->is_segment_overflow()) { // do reap ts if any of: // a. wait keyframe and got keyframe. // b. always reap when not wait keyframe. if (!muxer->wait_keyframe() || frame->frame_type == SrsVideoAvcFrameTypeKeyFrame) { // reap the segment, which will also flush the video. if ((err = reap_segment()) != srs_success) { return srs_error_wrap(err, "hls: reap segment"); } } } // flush video when got one if ((err = muxer->flush_video(tsmc)) != srs_success) { return srs_error_wrap(err, "hls: flush video"); } return err; } srs_error_t SrsHlsController::reap_segment() { srs_error_t err = srs_success; // TODO: flush audio before or after segment? // TODO: fresh segment begin with audio or video? // close current ts. if ((err = muxer->segment_close()) != srs_success) { // When close segment error, we must reopen it for next packet to write. srs_error_t r0 = muxer->segment_open(); if (r0 != srs_success) { srs_warn("close segment err %s", srs_error_desc(r0).c_str()); srs_freep(r0); } return srs_error_wrap(err, "hls: segment close"); } // open new ts. if ((err = muxer->segment_open()) != srs_success) { return srs_error_wrap(err, "hls: segment open"); } // segment open, flush video first. if ((err = muxer->flush_video(tsmc)) != srs_success) { return srs_error_wrap(err, "hls: flush video"); } // segment open, flush the audio. // @see: ngx_rtmp_hls_open_fragment /* start fragment with audio to make iPhone happy */ if ((err = muxer->flush_audio(tsmc)) != srs_success) { return srs_error_wrap(err, "hls: flush audio"); } return err; } SrsHls::SrsHls() { req = NULL; hub = NULL; enabled = false; disposable = false; last_update_time = 0; hls_dts_directly = false; previous_audio_dts = 0; aac_samples = 0; jitter = new SrsRtmpJitter(); controller = new SrsHlsController(); pprint = SrsPithyPrint::create_hls(); } SrsHls::~SrsHls() { srs_freep(jitter); srs_freep(controller); srs_freep(pprint); } void SrsHls::dispose() { if (enabled) { on_unpublish(); } // Ignore when hls_dispose disabled. // @see https://github.com/ossrs/srs/issues/865 srs_utime_t hls_dispose = _srs_config->get_hls_dispose(req->vhost); if (!hls_dispose) { return; } controller->dispose(); } srs_error_t SrsHls::cycle() { srs_error_t err = srs_success; if (last_update_time <= 0) { last_update_time = srs_get_system_time(); } if (!req) { return err; } srs_utime_t hls_dispose = _srs_config->get_hls_dispose(req->vhost); if (hls_dispose <= 0) { return err; } if (srs_get_system_time() - last_update_time <= hls_dispose) { return err; } last_update_time = srs_get_system_time(); if (!disposable) { return err; } disposable = false; srs_trace("hls cycle to dispose hls %s, timeout=%dms", req->get_stream_url().c_str(), hls_dispose); dispose(); return err; } srs_error_t SrsHls::initialize(SrsOriginHub* h, SrsRequest* r) { srs_error_t err = srs_success; hub = h; req = r; if ((err = controller->initialize()) != srs_success) { return srs_error_wrap(err, "controller initialize"); } return err; } srs_error_t SrsHls::on_publish() { srs_error_t err = srs_success; // update the hls time, for hls_dispose. last_update_time = srs_get_system_time(); // support multiple publish. if (enabled) { return err; } if (!_srs_config->get_hls_enabled(req->vhost)) { return err; } if ((err = controller->on_publish(req)) != srs_success) { return srs_error_wrap(err, "hls: on publish"); } // If enabled, directly turn FLV timestamp to TS DTS. // @remark It'll be reloaded automatically, because the origin hub will republish while reloading. hls_dts_directly = _srs_config->get_vhost_hls_dts_directly(req->vhost); // if enabled, open the muxer. enabled = true; // ok, the hls can be dispose, or need to be dispose. disposable = true; return err; } void SrsHls::on_unpublish() { srs_error_t err = srs_success; // support multiple unpublish. if (!enabled) { return; } if ((err = controller->on_unpublish()) != srs_success) { srs_warn("hls: ignore unpublish failed %s", srs_error_desc(err).c_str()); srs_freep(err); } enabled = false; } srs_error_t SrsHls::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format) { srs_error_t err = srs_success; if (!enabled) { return err; } // Ignore if no format->acodec, it means the codec is not parsed, or unknown codec. // @issue https://github.com/ossrs/srs/issues/1506#issuecomment-562079474 if (!format->acodec) { return err; } // update the hls time, for hls_dispose. last_update_time = srs_get_system_time(); SrsSharedPtrMessage* audio = shared_audio->copy(); SrsAutoFree(SrsSharedPtrMessage, audio); // ts support audio codec: aac/mp3 SrsAudioCodecId acodec = format->acodec->id; if (acodec != SrsAudioCodecIdAAC && acodec != SrsAudioCodecIdMP3) { return err; } // ignore sequence header srs_assert(format->audio); if (acodec == SrsAudioCodecIdAAC && format->audio->aac_packet_type == SrsAudioAacFrameTraitSequenceHeader) { return controller->on_sequence_header(); } // TODO: FIXME: config the jitter of HLS. if ((err = jitter->correct(audio, SrsRtmpJitterAlgorithmOFF)) != srs_success) { return srs_error_wrap(err, "hls: jitter"); } // Reset the aac samples counter when DTS jitter. if (previous_audio_dts > audio->timestamp) { previous_audio_dts = audio->timestamp; aac_samples = 0; } // The diff duration in ms between two FLV audio packets. int diff = ::abs((int)(audio->timestamp - previous_audio_dts)); previous_audio_dts = audio->timestamp; // Guess the number of samples for each AAC frame. // If samples is 1024, the sample-rate is 8000HZ, the diff should be 1024/8000s=128ms. // If samples is 1024, the sample-rate is 44100HZ, the diff should be 1024/44100s=23ms. // If samples is 2048, the sample-rate is 44100HZ, the diff should be 2048/44100s=46ms. int nb_samples_per_frame = 0; int guessNumberOfSamples = diff * srs_flv_srates[format->acodec->sound_rate] / 1000; if (guessNumberOfSamples > 0) { if (guessNumberOfSamples < 960) { nb_samples_per_frame = 960; } else if (guessNumberOfSamples < 1536) { nb_samples_per_frame = 1024; } else if (guessNumberOfSamples < 3072) { nb_samples_per_frame = 2048; } else { nb_samples_per_frame = 4096; } } // Recalc the DTS by the samples of AAC. aac_samples += nb_samples_per_frame; int64_t dts = 90000 * aac_samples / srs_flv_srates[format->acodec->sound_rate]; // If directly turn FLV timestamp, overwrite the guessed DTS. // @doc https://github.com/ossrs/srs/issues/1506#issuecomment-562063095 if (hls_dts_directly) { dts = audio->timestamp * 90; } if ((err = controller->write_audio(format->audio, dts)) != srs_success) { return srs_error_wrap(err, "hls: write audio"); } return err; } srs_error_t SrsHls::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format) { srs_error_t err = srs_success; if (!enabled) { return err; } // Ignore if no format->vcodec, it means the codec is not parsed, or unknown codec. // @issue https://github.com/ossrs/srs/issues/1506#issuecomment-562079474 if (!format->vcodec) { return err; } // update the hls time, for hls_dispose. last_update_time = srs_get_system_time(); SrsSharedPtrMessage* video = shared_video->copy(); SrsAutoFree(SrsSharedPtrMessage, video); // ignore info frame, // @see https://github.com/ossrs/srs/issues/288#issuecomment-69863909 srs_assert(format->video); if (format->video->frame_type == SrsVideoAvcFrameTypeVideoInfoFrame) { return err; } srs_assert(format->vcodec); if (format->vcodec->id != SrsVideoCodecIdAVC) { return err; } // ignore sequence header if (format->video->avc_packet_type == SrsVideoAvcFrameTraitSequenceHeader) { return controller->on_sequence_header(); } // TODO: FIXME: config the jitter of HLS. if ((err = jitter->correct(video, SrsRtmpJitterAlgorithmOFF)) != srs_success) { return srs_error_wrap(err, "hls: jitter"); } int64_t dts = video->timestamp * 90; if ((err = controller->write_video(format->video, dts)) != srs_success) { return srs_error_wrap(err, "hls: write video"); } // pithy print message. hls_show_mux_log(); return err; } void SrsHls::hls_show_mux_log() { pprint->elapse(); if (!pprint->can_print()) { return; } // the run time is not equals to stream time, // @see: https://github.com/ossrs/srs/issues/81#issuecomment-48100994 // it's ok. srs_trace("-> " SRS_CONSTS_LOG_HLS " time=%dms, sno=%d, ts=%s, dur=%.2f, dva=%dp", pprint->age(), controller->sequence_no(), controller->ts_url().c_str(), srsu2msi(controller->duration()), controller->deviation()); }