mirror of
https://github.com/ossrs/srs.git
synced 2025-03-09 15:49:59 +00:00
support hls ingest, fix the bugs.
This commit is contained in:
winlin
parent
2239e9f1fd
commit
ba6736839b
3 changed files with 234 additions and 80 deletions
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@ -169,6 +169,23 @@ int SrsTsMessage::stream_number()
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return -1;
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}
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SrsTsMessage* SrsTsMessage::detach()
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{
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// @remark the packet cannot be used, but channel is ok.
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SrsTsMessage* cp = new SrsTsMessage(channel, NULL);
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cp->start_pts = start_pts;
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cp->write_pcr = write_pcr;
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cp->is_discontinuity = is_discontinuity;
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cp->dts = dts;
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cp->pts = pts;
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cp->sid = sid;
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cp->PES_packet_length = PES_packet_length;
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cp->continuity_counter = continuity_counter;
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cp->payload = payload;
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payload = NULL;
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return cp;
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}
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ISrsTsHandler::ISrsTsHandler()
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{
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}
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@ -309,6 +309,13 @@ public:
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* @return the stream number for audio/video; otherwise, -1.
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*/
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virtual int stream_number();
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public:
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/**
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* detach the ts message,
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* for user maybe need to parse the message by queue.
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* @remark we always use the payload of original message.
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*/
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virtual SrsTsMessage* detach();
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};
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/**
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@ -23,8 +23,10 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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#include <srs_core.hpp>
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#include <stdlib.h>
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#include <string>
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#include <vector>
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#include <map>
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using namespace std;
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#include <srs_kernel_error.hpp>
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@ -148,7 +150,7 @@ private:
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dirty = false;
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}
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int fetch(std::string m3u8, SrsHttpClient* client);
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int fetch(std::string m3u8);
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};
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private:
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SrsHttpUri* in_hls;
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@ -196,7 +198,7 @@ private:
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/**
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* fetch all ts body.
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*/
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virtual void fetch_all_ts(bool fresh_m3u8, SrsHttpClient* client);
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virtual void fetch_all_ts(bool fresh_m3u8);
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/**
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* remove all ts which is dirty.
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*/
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@ -209,6 +211,7 @@ int SrsIngestSrsInput::connect()
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int64_t now = srs_update_system_time_ms();
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if (now < next_connect_time) {
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srs_trace("input hls wait for %dms", next_connect_time - now);
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st_usleep((next_connect_time - now) * 1000);
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}
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@ -328,7 +331,7 @@ int SrsIngestSrsInput::connect()
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}
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// fetch all ts.
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fetch_all_ts(fresh_m3u8, &client);
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fetch_all_ts(fresh_m3u8);
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// remove all dirty ts.
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remove_dirty();
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@ -344,12 +347,19 @@ int SrsIngestSrsInput::parse(ISrsTsHandler* handler)
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for (int i = 0; i < (int)pieces.size(); i++) {
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SrsTsPiece* tp = pieces.at(i);
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// sent only once.
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if (tp->sent) {
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continue;
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}
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tp->sent = true;
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if (tp->body.empty()) {
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continue;
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}
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srs_trace("proxy the ts to rtmp, ts=%s, duration=%.2f", tp->url.c_str(), tp->duration);
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// use stream to parse ts packet.
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int nb_packet = (int)tp->body.length() / SRS_TS_PACKET_SIZE;
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for (int i = 0; i < nb_packet; i++) {
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@ -360,6 +370,12 @@ int SrsIngestSrsInput::parse(ISrsTsHandler* handler)
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// process each ts packet
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if ((ret = context->decode(stream, handler)) != ERROR_SUCCESS) {
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// when peer closed, must interrupt parse and reconnect.
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if (srs_is_client_gracefully_close(ret)) {
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srs_warn("interrupt parse for peer closed. ret=%d", ret);
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return ret;
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}
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srs_warn("mpegts: ignore parse ts packet failed. ret=%d", ret);
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continue;
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}
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@ -392,7 +408,7 @@ void SrsIngestSrsInput::dirty_all_ts()
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}
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}
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void SrsIngestSrsInput::fetch_all_ts(bool fresh_m3u8, SrsHttpClient* client)
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void SrsIngestSrsInput::fetch_all_ts(bool fresh_m3u8)
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{
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int ret = ERROR_SUCCESS;
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@ -410,17 +426,16 @@ void SrsIngestSrsInput::fetch_all_ts(bool fresh_m3u8, SrsHttpClient* client)
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continue;
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}
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if ((ret = tp->fetch(in_hls->get_url(), client)) != ERROR_SUCCESS) {
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if ((ret = tp->fetch(in_hls->get_url())) != ERROR_SUCCESS) {
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srs_warn("ignore ts %s for error. ret=%d", tp->url.c_str(), ret);
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tp->skip = true;
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continue;
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}
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// set the next connect time.
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if (next_connect_time <= 0) {
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next_connect_time = srs_update_system_time_ms();
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// only wait for a duration of last piece.
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if (i == pieces.size() - 1) {
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next_connect_time = srs_update_system_time_ms() + (int)tp->duration * 1000;
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}
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next_connect_time += (int)tp->duration * 1000;
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}
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}
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@ -432,6 +447,7 @@ void SrsIngestSrsInput::remove_dirty()
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SrsTsPiece* tp = *it;
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if (tp->dirty) {
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srs_trace("erase dirty ts, url=%s, duration=%.2f", tp->url.c_str(), tp->duration);
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srs_freep(tp);
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it = pieces.erase(it);
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} else {
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@ -440,7 +456,7 @@ void SrsIngestSrsInput::remove_dirty()
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}
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}
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int SrsIngestSrsInput::SrsTsPiece::fetch(string m3u8, SrsHttpClient* client)
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int SrsIngestSrsInput::SrsTsPiece::fetch(string m3u8)
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{
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int ret = ERROR_SUCCESS;
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@ -450,8 +466,7 @@ int SrsIngestSrsInput::SrsTsPiece::fetch(string m3u8, SrsHttpClient* client)
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size_t pos = string::npos;
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bool use_abs_client = false;
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SrsHttpClient abs_client;
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SrsHttpClient client;
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std::string ts_url = url;
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if (!srs_string_starts_with(ts_url, "http://")) {
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@ -460,10 +475,6 @@ int SrsIngestSrsInput::SrsTsPiece::fetch(string m3u8, SrsHttpClient* client)
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baseurl = m3u8.substr(0, pos);
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}
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ts_url = baseurl + "/" + url;
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// use fresh client for absolute url.
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client = &abs_client;
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use_abs_client = true;
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}
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SrsHttpUri uri;
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@ -472,12 +483,12 @@ int SrsIngestSrsInput::SrsTsPiece::fetch(string m3u8, SrsHttpClient* client)
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}
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// initialize the fresh http client.
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if (use_abs_client && (ret = client->initialize(uri.get_host(), uri.get_port()) != ERROR_SUCCESS)) {
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if ((ret = client.initialize(uri.get_host(), uri.get_port()) != ERROR_SUCCESS)) {
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return ret;
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}
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SrsHttpMessage* msg = NULL;
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if ((ret = client->get(uri.get_path(), "", &msg)) != ERROR_SUCCESS) {
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if ((ret = client.get(uri.get_path(), "", &msg)) != ERROR_SUCCESS) {
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srs_error("HTTP GET %s failed. ret=%d", uri.get_url(), ret);
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return ret;
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}
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@ -500,6 +511,9 @@ class SrsIngestSrsOutput : public ISrsTsHandler
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{
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private:
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SrsHttpUri* out_rtmp;
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private:
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bool disconnected;
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std::multimap<int64_t, SrsTsMessage*> queue;
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private:
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SrsRequest* req;
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st_netfd_t stfd;
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@ -519,6 +533,7 @@ private:
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public:
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SrsIngestSrsOutput(SrsHttpUri* rtmp) {
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out_rtmp = rtmp;
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disconnected = false;
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req = NULL;
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io = NULL;
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@ -537,14 +552,22 @@ public:
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srs_freep(avc);
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srs_freep(aac);
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std::multimap<int64_t, SrsTsMessage*>::iterator it;
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for (it = queue.begin(); it != queue.end(); ++it) {
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SrsTsMessage* msg = it->second;
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srs_freep(msg);
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}
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queue.clear();
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}
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// interface ISrsTsHandler
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public:
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virtual int on_ts_message(SrsTsMessage* msg);
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private:
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virtual int parse_message_queue();
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virtual int on_ts_video(SrsTsMessage* msg, SrsStream* avs);
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virtual int write_h264_sps_pps(u_int32_t dts, u_int32_t pts);
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virtual int write_h264_ipb_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts);
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virtual int write_h264_ipb_frame(std::string ibps, SrsCodecVideoAVCFrame frame_type, u_int32_t dts, u_int32_t pts);
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virtual int on_ts_audio(SrsTsMessage* msg, SrsStream* avs);
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virtual int write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t dts);
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private:
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@ -554,6 +577,10 @@ public:
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* connect to output rtmp server.
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*/
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virtual int connect();
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/**
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* flush the message queue when all ts parsed.
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*/
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virtual int flush_message_queue();
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private:
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virtual int connect_app(std::string ep_server, std::string ep_port);
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// close the connected io and rtmp to ready to be re-connect.
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@ -601,7 +628,7 @@ int SrsIngestSrsOutput::on_ts_message(SrsTsMessage* msg)
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// 14496-2 video stream number xxxx
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// ((stream_id >> 4) & 0x0f) == SrsTsPESStreamIdVideo
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srs_trace("<- "SRS_CONSTS_LOG_STREAM_CASTER" mpegts: got %s stream=%s, dts=%"PRId64", pts=%"PRId64", size=%d, us=%d, cc=%d, sid=%#x(%s-%d)",
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srs_info("<- "SRS_CONSTS_LOG_STREAM_CASTER" mpegts: got %s stream=%s, dts=%"PRId64", pts=%"PRId64", size=%d, us=%d, cc=%d, sid=%#x(%s-%d)",
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(msg->channel->apply == SrsTsPidApplyVideo)? "Video":"Audio", srs_ts_stream2string(msg->channel->stream).c_str(),
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msg->dts, msg->pts, msg->payload->length(), msg->packet->payload_unit_start_indicator, msg->continuity_counter, msg->sid,
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msg->is_audio()? "A":msg->is_video()? "V":"N", msg->stream_number());
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@ -621,22 +648,107 @@ int SrsIngestSrsOutput::on_ts_message(SrsTsMessage* msg)
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return ret;
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}
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// parse the stream.
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SrsStream avs;
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if ((ret = avs.initialize(msg->payload->bytes(), msg->payload->length())) != ERROR_SUCCESS) {
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srs_error("mpegts: initialize av stream failed. ret=%d", ret);
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// we must use queue to cache the msg, then parse it if possible.
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queue.insert(std::make_pair(msg->dts, msg->detach()));
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if ((ret = parse_message_queue()) != ERROR_SUCCESS) {
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// when peer closed, close the output and reconnect.
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if (srs_is_client_gracefully_close(ret)) {
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close();
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}
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return ret;
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}
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// publish audio or video.
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if (msg->channel->stream == SrsTsStreamVideoH264) {
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return on_ts_video(msg, &avs);
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}
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if (msg->channel->stream == SrsTsStreamAudioAAC) {
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return on_ts_audio(msg, &avs);
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return ret;
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}
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int SrsIngestSrsOutput::parse_message_queue()
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{
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int ret = ERROR_SUCCESS;
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int nb_videos = 0;
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int nb_audios = 0;
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std::multimap<int64_t, SrsTsMessage*>::iterator it;
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for (it = queue.begin(); it != queue.end(); ++it) {
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SrsTsMessage* msg = it->second;
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// publish audio or video.
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if (msg->channel->stream == SrsTsStreamVideoH264) {
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nb_videos++;
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} else {
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nb_audios++;
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}
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}
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// always wait 2+ videos, to left one video in the queue.
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// TODO: FIXME: support pure audio hls.
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if (nb_videos <= 1) {
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return ret;
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}
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// parse messages util the last video.
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while (nb_videos > 1 && queue.size() > 0) {
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std::multimap<int64_t, SrsTsMessage*>::iterator it = queue.begin();
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SrsTsMessage* msg = it->second;
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if (msg->channel->stream == SrsTsStreamVideoH264) {
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nb_videos--;
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}
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queue.erase(it);
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// parse the stream.
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SrsStream avs;
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if ((ret = avs.initialize(msg->payload->bytes(), msg->payload->length())) != ERROR_SUCCESS) {
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srs_error("mpegts: initialize av stream failed. ret=%d", ret);
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return ret;
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}
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// publish audio or video.
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if (msg->channel->stream == SrsTsStreamVideoH264) {
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if ((ret = on_ts_video(msg, &avs)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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if (msg->channel->stream == SrsTsStreamAudioAAC) {
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if ((ret = on_ts_audio(msg, &avs)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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}
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return ret;
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}
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int SrsIngestSrsOutput::flush_message_queue()
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{
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int ret = ERROR_SUCCESS;
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// parse messages util the last video.
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while (!queue.empty()) {
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std::multimap<int64_t, SrsTsMessage*>::iterator it = queue.begin();
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SrsTsMessage* msg = it->second;
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queue.erase(it);
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// parse the stream.
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SrsStream avs;
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if ((ret = avs.initialize(msg->payload->bytes(), msg->payload->length())) != ERROR_SUCCESS) {
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srs_error("mpegts: initialize av stream failed. ret=%d", ret);
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return ret;
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}
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// publish audio or video.
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if (msg->channel->stream == SrsTsStreamVideoH264) {
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if ((ret = on_ts_video(msg, &avs)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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if (msg->channel->stream == SrsTsStreamAudioAAC) {
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if ((ret = on_ts_audio(msg, &avs)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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}
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// TODO: FIXME: implements it.
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return ret;
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}
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@ -644,18 +756,12 @@ int SrsIngestSrsOutput::on_ts_video(SrsTsMessage* msg, SrsStream* avs)
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{
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int ret = ERROR_SUCCESS;
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// ensure rtmp connected.
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if ((ret = connect()) != ERROR_SUCCESS) {
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return ret;
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}
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// ts tbn to flv tbn.
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u_int32_t dts = (u_int32_t)(msg->dts / 90);
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u_int32_t pts = (u_int32_t)(msg->dts / 90);
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// the whole ts pes video packet must be a flv frame packet.
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char* ibpframe = avs->data() + avs->pos();
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int ibpframe_size = avs->size() - avs->pos();
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std::string ibps;
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SrsCodecVideoAVCFrame frame_type = SrsCodecVideoAVCFrameInterFrame;
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// send each frame.
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while (!avs->empty()) {
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@ -665,10 +771,18 @@ int SrsIngestSrsOutput::on_ts_video(SrsTsMessage* msg, SrsStream* avs)
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return ret;
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}
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// ignore invalid frame,
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// * atleast 1bytes for SPS to decode the type
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// * ignore the auth bytes '09f0'
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if (frame_size <= 2) {
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// 5bits, 7.3.1 NAL unit syntax,
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// H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
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// 7: SPS, 8: PPS, 5: I Frame, 1: P Frame
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SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
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// for IDR frame, the frame is keyframe.
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if (nal_unit_type == SrsAvcNaluTypeIDR) {
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frame_type = SrsCodecVideoAVCFrameKeyFrame;
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}
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// ignore the nalu type aud(9)
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if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter) {
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continue;
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}
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@ -684,10 +798,6 @@ int SrsIngestSrsOutput::on_ts_video(SrsTsMessage* msg, SrsStream* avs)
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}
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h264_sps_changed = true;
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h264_sps = sps;
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if ((ret = write_h264_sps_pps(dts, pts)) != ERROR_SUCCESS) {
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||||
return ret;
|
||||
}
|
||||
continue;
|
||||
}
|
||||
|
||||
|
|
@ -703,27 +813,45 @@ int SrsIngestSrsOutput::on_ts_video(SrsTsMessage* msg, SrsStream* avs)
|
|||
}
|
||||
h264_pps_changed = true;
|
||||
h264_pps = pps;
|
||||
|
||||
if ((ret = write_h264_sps_pps(dts, pts)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
continue;
|
||||
}
|
||||
|
||||
break;
|
||||
// ibp frame.
|
||||
std::string ibp;
|
||||
if ((ret = avc->mux_ipb_frame(frame, frame_size, ibp)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
ibps.append(ibp);
|
||||
}
|
||||
|
||||
// ibp frame.
|
||||
srs_info("mpegts: demux avc ibp frame size=%d, dts=%d", ibpframe_size, dts);
|
||||
return write_h264_ipb_frame(ibpframe, ibpframe_size, dts, pts);
|
||||
if ((ret = write_h264_sps_pps(dts, pts)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
if ((ret = write_h264_ipb_frame(ibps, frame_type, dts, pts)) != ERROR_SUCCESS) {
|
||||
// drop the ts message.
|
||||
if (ret == ERROR_H264_DROP_BEFORE_SPS_PPS) {
|
||||
return ERROR_SUCCESS;
|
||||
}
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsIngestSrsOutput::write_h264_sps_pps(u_int32_t dts, u_int32_t pts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// only send when both sps and pps changed.
|
||||
if (!h264_sps_changed || !h264_pps_changed) {
|
||||
// when sps or pps changed, update the sequence header,
|
||||
// for the pps maybe not changed while sps changed.
|
||||
// so, we must check when each video ts message frame parsed.
|
||||
if (h264_sps_pps_sent && !h264_sps_changed && !h264_pps_changed) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// when not got sps/pps, wait.
|
||||
if (h264_pps.empty() || h264_sps.empty()) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
|
|
@ -752,11 +880,12 @@ int SrsIngestSrsOutput::write_h264_sps_pps(u_int32_t dts, u_int32_t pts)
|
|||
h264_sps_changed = false;
|
||||
h264_pps_changed = false;
|
||||
h264_sps_pps_sent = true;
|
||||
srs_trace("hls: h264 sps/pps sent, sps=%dB, pps=%dB", h264_sps.length(), h264_pps.length());
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
int SrsIngestSrsOutput::write_h264_ipb_frame(char* frame, int frame_size, u_int32_t dts, u_int32_t pts)
|
||||
int SrsIngestSrsOutput::write_h264_ipb_frame(string ibps, SrsCodecVideoAVCFrame frame_type, u_int32_t dts, u_int32_t pts)
|
||||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
|
|
@ -766,26 +895,10 @@ int SrsIngestSrsOutput::write_h264_ipb_frame(char* frame, int frame_size, u_int3
|
|||
return ERROR_H264_DROP_BEFORE_SPS_PPS;
|
||||
}
|
||||
|
||||
// 5bits, 7.3.1 NAL unit syntax,
|
||||
// H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
|
||||
// 7: SPS, 8: PPS, 5: I Frame, 1: P Frame
|
||||
SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
|
||||
|
||||
// for IDR frame, the frame is keyframe.
|
||||
SrsCodecVideoAVCFrame frame_type = SrsCodecVideoAVCFrameInterFrame;
|
||||
if (nal_unit_type == SrsAvcNaluTypeIDR) {
|
||||
frame_type = SrsCodecVideoAVCFrameKeyFrame;
|
||||
}
|
||||
|
||||
std::string ibp;
|
||||
if ((ret = avc->mux_ipb_frame(frame, frame_size, ibp)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
int8_t avc_packet_type = SrsCodecVideoAVCTypeNALU;
|
||||
char* flv = NULL;
|
||||
int nb_flv = 0;
|
||||
if ((ret = avc->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != ERROR_SUCCESS) {
|
||||
if ((ret = avc->mux_avc2flv(ibps, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
|
|
@ -798,14 +911,17 @@ int SrsIngestSrsOutput::on_ts_audio(SrsTsMessage* msg, SrsStream* avs)
|
|||
{
|
||||
int ret = ERROR_SUCCESS;
|
||||
|
||||
// ensure rtmp connected.
|
||||
if ((ret = connect()) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// ts tbn to flv tbn.
|
||||
u_int32_t dts = (u_int32_t)(msg->dts / 90);
|
||||
|
||||
// got the next video to calc the delta duration for each audio.
|
||||
u_int32_t duration = 0;
|
||||
if (!queue.empty()) {
|
||||
SrsTsMessage* nm = queue.begin()->second;
|
||||
duration = (u_int32_t)(srs_max(0, nm->dts - msg->dts) / 90);
|
||||
}
|
||||
u_int32_t max_dts = dts + duration;
|
||||
|
||||
// send each frame.
|
||||
while (!avs->empty()) {
|
||||
char* frame = NULL;
|
||||
|
|
@ -842,6 +958,10 @@ int SrsIngestSrsOutput::on_ts_audio(SrsTsMessage* msg, SrsStream* avs)
|
|||
if ((ret = write_audio_raw_frame(frame, frame_size, &codec, dts)) != ERROR_SUCCESS) {
|
||||
return ret;
|
||||
}
|
||||
|
||||
// calc the delta of dts, when previous frame output.
|
||||
u_int32_t delta = duration / (msg->payload->length() / frame_size);
|
||||
dts = (u_int32_t)(srs_min(max_dts, dts + delta));
|
||||
}
|
||||
|
||||
return ret;
|
||||
|
|
@ -890,6 +1010,8 @@ int SrsIngestSrsOutput::connect()
|
|||
return ret;
|
||||
}
|
||||
|
||||
srs_trace("connect output=%s", out_rtmp->get_url());
|
||||
|
||||
// parse uri
|
||||
if (!req) {
|
||||
req = new SrsRequest();
|
||||
|
|
@ -996,6 +1118,9 @@ int SrsIngestSrsOutput::connect_app(string ep_server, string ep_port)
|
|||
|
||||
void SrsIngestSrsOutput::close()
|
||||
{
|
||||
srs_trace("close output=%s", out_rtmp->get_url());
|
||||
h264_sps_pps_sent = false;
|
||||
|
||||
srs_freep(client);
|
||||
srs_freep(io);
|
||||
srs_freep(req);
|
||||
|
|
@ -1035,6 +1160,11 @@ public:
|
|||
return ret;
|
||||
}
|
||||
|
||||
if ((ret = oc->flush_message_queue()) != ERROR_SUCCESS) {
|
||||
srs_warn("flush oc message failed. ret=%d", ret);
|
||||
return ret;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
};
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue