mirror of
https://github.com/ossrs/srs.git
synced 2025-03-09 15:49:59 +00:00
1. Modify rtc.conf to support Bframe discard.
2. Rename srs_app_rtp.cpp to srs_app_rtc.cpp
This commit is contained in:
xiaozhihong
parent
68ad006b73
commit
fa700dad64
10 changed files with 129 additions and 69 deletions
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@ -309,7 +309,7 @@ srt_server {
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#############################################################################################
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# WebRTC server section
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#############################################################################################
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rtc {
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rtc_server {
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# Whether enable WebRTC server.
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# default: off
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enabled on;
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@ -18,7 +18,7 @@ http_api {
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stats {
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network 0;
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}
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rtc {
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rtc_server {
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enabled on;
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# Listen at udp://8000
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listen 8000;
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@ -32,5 +32,9 @@ rtc {
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}
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vhost __defaultVhost__ {
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rtc {
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enabled on;
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bframe discard;
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}
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}
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2
trunk/configure
vendored
2
trunk/configure
vendored
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@ -256,7 +256,7 @@ if [ $SRS_EXPORT_LIBRTMP_PROJECT = NO ]; then
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"srs_app_ingest" "srs_app_ffmpeg" "srs_app_utility" "srs_app_edge"
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"srs_app_heartbeat" "srs_app_empty" "srs_app_http_client" "srs_app_http_static"
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"srs_app_recv_thread" "srs_app_security" "srs_app_statistic" "srs_app_hds"
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"srs_app_mpegts_udp" "srs_app_rtp" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
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"srs_app_mpegts_udp" "srs_app_rtc" "srs_app_rtc_conn" "srs_app_dtls" "srs_app_rtsp" "srs_app_listener" "srs_app_async_call"
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"srs_app_caster_flv" "srs_app_process" "srs_app_ng_exec"
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"srs_app_hourglass" "srs_app_dash" "srs_app_fragment" "srs_app_dvr"
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"srs_app_coworkers" "srs_app_hybrid")
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@ -3486,7 +3486,7 @@ srs_error_t SrsConfig::check_normal_config()
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&& n != "srs_log_tank" && n != "srs_log_level" && n != "srs_log_file"
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&& n != "max_connections" && n != "daemon" && n != "heartbeat"
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&& n != "http_api" && n != "stats" && n != "vhost" && n != "pithy_print_ms"
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&& n != "http_server" && n != "stream_caster" && n != "rtc" && n != "srt_server"
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&& n != "http_server" && n != "stream_caster" && n != "rtc_server" && n != "srt_server"
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&& n != "utc_time" && n != "work_dir" && n != "asprocess"
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&& n != "ff_log_level" && n != "grace_final_wait" && n != "force_grace_quit"
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&& n != "grace_start_wait" && n != "empty_ip_ok" && n != "disable_daemon_for_docker"
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@ -3673,7 +3673,7 @@ srs_error_t SrsConfig::check_normal_config()
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&& n != "play" && n != "publish" && n != "cluster"
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&& n != "security" && n != "http_remux" && n != "dash"
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&& n != "http_static" && n != "hds" && n != "exec"
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&& n != "in_ack_size" && n != "out_ack_size") {
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&& n != "in_ack_size" && n != "out_ack_size" && n != "rtc") {
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return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.%s", n.c_str());
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}
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// for each sub directives of vhost.
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@ -3819,6 +3819,13 @@ srs_error_t SrsConfig::check_normal_config()
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return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.bandcheck.%s of %s", m.c_str(), vhost->arg0().c_str());
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}
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}
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} else if (n == "rtc") {
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for (int j = 0; j < (int)conf->directives.size(); j++) {
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string m = conf->at(j)->name;
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if (m != "enabled" && m != "bframe") {
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return srs_error_new(ERROR_SYSTEM_CONFIG_INVALID, "illegal vhost.bandcheck.%s of %s", m.c_str(), vhost->arg0().c_str());
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}
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}
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}
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}
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}
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@ -4266,13 +4273,13 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf)
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return ::atoi(conf->arg0().c_str());
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}
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int SrsConfig::get_rtc_enabled()
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int SrsConfig::get_rtc_server_enabled()
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{
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SrsConfDirective* conf = root->get("rtc");
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return get_rtc_enabled(conf);
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SrsConfDirective* conf = root->get("rtc_server");
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return get_rtc_server_enabled(conf);
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}
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bool SrsConfig::get_rtc_enabled(SrsConfDirective* conf)
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bool SrsConfig::get_rtc_server_enabled(SrsConfDirective* conf)
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{
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static bool DEFAULT = false;
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@ -4288,11 +4295,11 @@ bool SrsConfig::get_rtc_enabled(SrsConfDirective* conf)
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return SRS_CONF_PERFER_FALSE(conf->arg0());
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}
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int SrsConfig::get_rtc_listen()
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int SrsConfig::get_rtc_server_listen()
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{
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static int DEFAULT = 8000;
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SrsConfDirective* conf = root->get("rtc");
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SrsConfDirective* conf = root->get("rtc_server");
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if (!conf) {
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return DEFAULT;
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}
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@ -4305,11 +4312,11 @@ int SrsConfig::get_rtc_listen()
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return ::atoi(conf->arg0().c_str());
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}
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std::string SrsConfig::get_rtc_candidates()
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std::string SrsConfig::get_rtc_server_candidates()
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{
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static string DEFAULT = "*";
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SrsConfDirective* conf = root->get("rtc");
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SrsConfDirective* conf = root->get("rtc_server");
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if (!conf) {
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return DEFAULT;
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}
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@ -4332,6 +4339,48 @@ std::string SrsConfig::get_rtc_candidates()
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return (conf->arg0().c_str());
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}
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SrsConfDirective* SrsConfig::get_rtc(string vhost)
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{
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SrsConfDirective* conf = get_vhost(vhost);
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return conf? conf->get("rtc") : NULL;
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}
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bool SrsConfig::get_rtc_enabled(string vhost)
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{
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static bool DEFAULT = false;
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SrsConfDirective* conf = get_rtc(vhost);
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if (!conf) {
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return DEFAULT;
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}
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conf = conf->get("enabled");
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if (!conf || conf->arg0().empty()) {
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return DEFAULT;
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}
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return SRS_CONF_PERFER_FALSE(conf->arg0());
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}
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bool SrsConfig::get_rtc_bframe_discard(string vhost)
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{
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static bool DEFAULT = false;
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SrsConfDirective* conf = get_rtc(vhost);
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if (!conf) {
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return DEFAULT;
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}
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conf = conf->get("enabled");
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if (!conf || conf->arg0().empty()) {
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return DEFAULT;
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}
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return conf->arg0() == "discard";
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}
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SrsConfDirective* SrsConfig::get_vhost(string vhost, bool try_default_vhost)
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{
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srs_assert(root);
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@ -501,10 +501,14 @@ public:
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// rtc section
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public:
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virtual int get_rtc_enabled();
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virtual bool get_rtc_enabled(SrsConfDirective* conf);
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virtual int get_rtc_listen();
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virtual std::string get_rtc_candidates();
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virtual int get_rtc_server_enabled();
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virtual bool get_rtc_server_enabled(SrsConfDirective* conf);
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virtual int get_rtc_server_listen();
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virtual std::string get_rtc_server_candidates();
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SrsConfDirective* get_rtc(std::string vhost);
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bool get_rtc_enabled(std::string vhost);
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bool get_rtc_bframe_discard(std::string vhost);
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// vhost specified section
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public:
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@ -21,7 +21,7 @@
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* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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*/
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#include <srs_app_rtp.hpp>
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#include <srs_app_rtc.hpp>
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#include <sys/types.h>
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#include <sys/stat.h>
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@ -49,11 +49,13 @@ using namespace std;
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#include <srs_app_utility.hpp>
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#include <srs_app_http_hooks.hpp>
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#include <srs_protocol_format.hpp>
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#include <srs_rtmp_stack.hpp>
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#include <openssl/rand.h>
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SrsRtpMuxer::SrsRtpMuxer()
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{
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sequence = 0;
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discard_bframe = false;
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}
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SrsRtpMuxer::~SrsRtpMuxer()
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@ -99,11 +101,12 @@ srs_error_t SrsRtpMuxer::frame_to_packet(SrsSharedPtrMessage* shared_frame, SrsF
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}
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srs_verbose("nal_type=%d, slice type=%d", nal_type, slice_type);
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// TODO: Use config to determine how to process B frame
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if (slice_type == SrsAvcSliceTypeB || slice_type == SrsAvcSliceTypeB1) {
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if (discard_bframe) {
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continue;
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}
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}
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}
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if (sample.size <= max_payload_size) {
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if ((err = packet_single_nalu(shared_frame, format, &sample, rtp_packet_vec)) != srs_success) {
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@ -179,11 +182,8 @@ srs_error_t SrsRtpMuxer::packet_fu_a(SrsSharedPtrMessage* shared_frame, SrsForma
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p += packet_size;
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nb_left -= packet_size;
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srs_verbose("rtp fu-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u, rtp header=%s, payload=%s",
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sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType,
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srs_string_dumps_hex(stream->data(), 12).c_str(),
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srs_string_dumps_hex(stream->data() + 12, stream->pos() - 12).c_str());
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srs_verbose("rtp fu-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u",
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sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType);
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SrsRtpSharedPacket* rtp_shared_pkt = new SrsRtpSharedPacket();
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rtp_shared_pkt->create((shared_frame->timestamp * 90), sequence++, kVideoSSRC, kH264PayloadType, stream->data(), stream->pos());
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@ -224,11 +224,8 @@ srs_error_t SrsRtpMuxer::packet_single_nalu(SrsSharedPtrMessage* shared_frame, S
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stream->write_bytes(sample->bytes, sample->size);
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srs_verbose("sample=%s", srs_string_dumps_hex(sample->bytes, sample->size).c_str());
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srs_verbose("rtp single nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u, rtp header=%s, payload=%s",
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sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType,
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srs_string_dumps_hex(stream->data(), 12).c_str(),
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srs_string_dumps_hex(stream->data() + 12, stream->pos() - 12).c_str());
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srs_verbose("rtp single nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u",
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sample->size, sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType);
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SrsRtpSharedPacket* rtp_shared_pkt = new SrsRtpSharedPacket();
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rtp_shared_pkt->create((shared_frame->timestamp * 90), sequence++, kVideoSSRC, kH264PayloadType, stream->data(), stream->pos());
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@ -275,10 +272,8 @@ srs_error_t SrsRtpMuxer::packet_stap_a(const string &sps, const string& pps, Srs
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stream->write_2bytes(pps.size());
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stream->write_bytes((char*)pps.data(), pps.size());
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srs_verbose("rtp stap-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u, rtp header=%s, payload=%s",
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(sps.size() + pps.size()), sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType,
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srs_string_dumps_hex(stream->data(), 12).c_str(),
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srs_string_dumps_hex(stream->data() + 12, stream->pos() - 12).c_str());
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srs_verbose("rtp stap-a nalu, size=%u, seq=%u, timestamp=%lu, ssrc=%u, payloadtype=%u",
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(sps.size() + pps.size()), sequence, (shared_frame->timestamp * 90), kVideoSSRC, kH264PayloadType);
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SrsRtpSharedPacket* rtp_shared_pkt = new SrsRtpSharedPacket();
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rtp_shared_pkt->create((shared_frame->timestamp * 90), sequence++, kVideoSSRC, kH264PayloadType, stream->data(), stream->pos());
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@ -288,7 +283,7 @@ srs_error_t SrsRtpMuxer::packet_stap_a(const string &sps, const string& pps, Srs
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return err;
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}
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SrsRtp::SrsRtp()
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SrsRtc::SrsRtc()
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{
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req = NULL;
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hub = NULL;
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@ -298,12 +293,12 @@ SrsRtp::SrsRtp()
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last_update_time = 0;
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}
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SrsRtp::~SrsRtp()
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SrsRtc::~SrsRtc()
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{
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srs_freep(rtp_h264_muxer);
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}
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void SrsRtp::dispose()
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void SrsRtc::dispose()
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{
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if (enabled) {
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on_unpublish();
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@ -311,14 +306,14 @@ void SrsRtp::dispose()
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}
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// TODO: FIXME: Dead code?
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srs_error_t SrsRtp::cycle()
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srs_error_t SrsRtc::cycle()
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{
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srs_error_t err = srs_success;
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return err;
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}
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srs_error_t SrsRtp::initialize(SrsOriginHub* h, SrsRequest* r)
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srs_error_t SrsRtc::initialize(SrsOriginHub* h, SrsRequest* r)
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{
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srs_error_t err = srs_success;
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@ -326,11 +321,12 @@ srs_error_t SrsRtp::initialize(SrsOriginHub* h, SrsRequest* r)
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req = r;
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rtp_h264_muxer = new SrsRtpMuxer();
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rtp_h264_muxer->discard_bframe = _srs_config->get_rtc_bframe_discard(req->vhost);
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return err;
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}
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srs_error_t SrsRtp::on_publish()
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srs_error_t SrsRtc::on_publish()
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{
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srs_error_t err = srs_success;
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@ -342,6 +338,10 @@ srs_error_t SrsRtp::on_publish()
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return err;
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}
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if (!_srs_config->get_rtc_enabled(req->vhost)) {
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return err;
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}
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// if enabled, open the muxer.
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enabled = true;
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@ -351,7 +351,7 @@ srs_error_t SrsRtp::on_publish()
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return err;
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}
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void SrsRtp::on_unpublish()
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void SrsRtc::on_unpublish()
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{
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// support multiple unpublish.
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if (!enabled) {
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@ -361,7 +361,7 @@ void SrsRtp::on_unpublish()
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enabled = false;
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}
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srs_error_t SrsRtp::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format)
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srs_error_t SrsRtc::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* format)
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{
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srs_error_t err = srs_success;
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@ -391,7 +391,7 @@ srs_error_t SrsRtp::on_audio(SrsSharedPtrMessage* shared_audio, SrsFormat* forma
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return err;
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}
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srs_error_t SrsRtp::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format)
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srs_error_t SrsRtc::on_video(SrsSharedPtrMessage* shared_video, SrsFormat* format)
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{
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srs_error_t err = srs_success;
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@ -21,8 +21,8 @@
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* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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*/
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#ifndef SRS_APP_RTP_HPP
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#define SRS_APP_RTP_HPP
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#ifndef SRS_APP_RTC_HPP
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#define SRS_APP_RTC_HPP
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#include <srs_core.hpp>
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@ -53,12 +53,15 @@ const uint8_t kEnd = 0x40;
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// FIXME: ssrc can relate to source
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const uint32_t kVideoSSRC = 3233846889;
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// TODO: Define interface class like ISrsRtpMuxer to support SrsRtpOpusMuxer and so on.
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class SrsRtpMuxer
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{
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private:
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uint16_t sequence;
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std::string sps;
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std::string pps;
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public:
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bool discard_bframe;
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public:
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SrsRtpMuxer();
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virtual ~SrsRtpMuxer();
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@ -70,7 +73,7 @@ private:
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srs_error_t packet_stap_a(const std::string &sps, const std::string& pps, SrsSharedPtrMessage* shared_frame, std::vector<SrsRtpSharedPacket*>& rtp_packet_vec);
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};
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class SrsRtp
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class SrsRtc
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{
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private:
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SrsRequest* req;
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||||
|
|
@ -80,8 +83,8 @@ private:
|
|||
SrsRtpMuxer* rtp_h264_muxer;
|
||||
SrsOriginHub* hub;
|
||||
public:
|
||||
SrsRtp();
|
||||
virtual ~SrsRtp();
|
||||
SrsRtc();
|
||||
virtual ~SrsRtc();
|
||||
public:
|
||||
virtual void dispose();
|
||||
virtual srs_error_t cycle();
|
||||
|
|
@ -46,7 +46,7 @@ using namespace std;
|
|||
#include <srs_app_dtls.hpp>
|
||||
#include <srs_app_utility.hpp>
|
||||
#include <srs_app_config.hpp>
|
||||
#include <srs_app_rtp.hpp>
|
||||
#include <srs_app_rtc.hpp>
|
||||
#include <srs_app_source.hpp>
|
||||
#include <srs_app_server.hpp>
|
||||
#include <srs_service_utility.hpp>
|
||||
|
|
@ -101,7 +101,7 @@ std::vector<std::string> SrsCandidate::get_candidate_ips()
|
|||
{
|
||||
std::vector<std::string> candidate_ips;
|
||||
|
||||
string candidate = _srs_config->get_rtc_candidates();
|
||||
string candidate = _srs_config->get_rtc_server_candidates();
|
||||
if (candidate == "*" || candidate == "0.0.0.0") {
|
||||
std::vector<std::string> tmp = srs_get_local_ips();
|
||||
for (int i = 0; i < (int)tmp.size(); ++i) {
|
||||
|
|
@ -189,7 +189,7 @@ srs_error_t SrsSdp::encode(string& sdp_str)
|
|||
std::vector<string> candidate_ips = SrsCandidate::get_candidate_ips();
|
||||
for (int i = 0; i < (int)candidate_ips.size(); ++i) {
|
||||
ostringstream os;
|
||||
os << "a=candidate:10 1 udp 2115783679 " << candidate_ips[i] << " " << _srs_config->get_rtc_listen() <<" typ host generation 0\\r\\n";
|
||||
os << "a=candidate:10 1 udp 2115783679 " << candidate_ips[i] << " " << _srs_config->get_rtc_server_listen() <<" typ host generation 0\\r\\n";
|
||||
candidate_lines += os.str();
|
||||
}
|
||||
|
||||
|
|
@ -1179,11 +1179,11 @@ srs_error_t SrsRtcServer::listen_udp()
|
|||
{
|
||||
srs_error_t err = srs_success;
|
||||
|
||||
if (!_srs_config->get_rtc_enabled()) {
|
||||
if (!_srs_config->get_rtc_server_enabled()) {
|
||||
return err;
|
||||
}
|
||||
|
||||
int port = _srs_config->get_rtc_listen();
|
||||
int port = _srs_config->get_rtc_server_listen();
|
||||
if (port <= 0) {
|
||||
return srs_error_new(ERROR_RTC_PORT, "invalid port=%d", port);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -33,7 +33,7 @@ using namespace std;
|
|||
#include <srs_kernel_codec.hpp>
|
||||
#include <srs_kernel_rtp.hpp>
|
||||
#include <srs_app_hls.hpp>
|
||||
#include <srs_app_rtp.hpp>
|
||||
#include <srs_app_rtc.hpp>
|
||||
#include <srs_app_forward.hpp>
|
||||
#include <srs_app_config.hpp>
|
||||
#include <srs_app_encoder.hpp>
|
||||
|
|
@ -875,7 +875,7 @@ SrsOriginHub::SrsOriginHub()
|
|||
dash = new SrsDash();
|
||||
dvr = new SrsDvr();
|
||||
encoder = new SrsEncoder();
|
||||
rtp = new SrsRtp();
|
||||
rtc = new SrsRtc();
|
||||
#ifdef SRS_AUTO_HDS
|
||||
hds = new SrsHds();
|
||||
#endif
|
||||
|
|
@ -920,8 +920,8 @@ srs_error_t SrsOriginHub::initialize(SrsSource* s, SrsRequest* r)
|
|||
return srs_error_wrap(err, "format initialize");
|
||||
}
|
||||
|
||||
if ((err = rtp->initialize(this, req)) != srs_success) {
|
||||
return srs_error_wrap(err, "rtp initialize");
|
||||
if ((err = rtc->initialize(this, req)) != srs_success) {
|
||||
return srs_error_wrap(err, "rtc initialize");
|
||||
}
|
||||
|
||||
if ((err = hls->initialize(this, req)) != srs_success) {
|
||||
|
|
@ -1022,10 +1022,10 @@ srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
|
|||
srs_flv_srates[c->sound_rate]);
|
||||
}
|
||||
|
||||
if ((err = rtp->on_audio(msg, format)) != srs_success) {
|
||||
srs_warn("rtp: ignore audio error %s", srs_error_desc(err).c_str());
|
||||
if ((err = rtc->on_audio(msg, format)) != srs_success) {
|
||||
srs_warn("rtc: ignore audio error %s", srs_error_desc(err).c_str());
|
||||
srs_error_reset(err);
|
||||
rtp->on_unpublish();
|
||||
rtc->on_unpublish();
|
||||
}
|
||||
|
||||
if ((err = hls->on_audio(msg, format)) != srs_success) {
|
||||
|
|
@ -1122,11 +1122,11 @@ srs_error_t SrsOriginHub::on_video(SrsSharedPtrMessage* shared_video, bool is_se
|
|||
}
|
||||
|
||||
// Parse RTMP message to RTP packets, in FU-A if too large.
|
||||
if ((err = rtp->on_video(msg, format)) != srs_success) {
|
||||
if ((err = rtc->on_video(msg, format)) != srs_success) {
|
||||
// TODO: We should support more strategies.
|
||||
srs_warn("rtp: ignore video error %s", srs_error_desc(err).c_str());
|
||||
srs_warn("rtc: ignore video error %s", srs_error_desc(err).c_str());
|
||||
srs_error_reset(err);
|
||||
rtp->on_unpublish();
|
||||
rtc->on_unpublish();
|
||||
}
|
||||
|
||||
// TODO: FIXME: Refactor to move to rtp?
|
||||
|
|
@ -1201,8 +1201,8 @@ srs_error_t SrsOriginHub::on_publish()
|
|||
return srs_error_wrap(err, "encoder publish");
|
||||
}
|
||||
|
||||
if ((err = rtp->on_publish()) != srs_success) {
|
||||
return srs_error_wrap(err, "rtp publish");
|
||||
if ((err = rtc->on_publish()) != srs_success) {
|
||||
return srs_error_wrap(err, "rtc publish");
|
||||
}
|
||||
|
||||
if ((err = hls->on_publish()) != srs_success) {
|
||||
|
|
@ -1242,7 +1242,7 @@ void SrsOriginHub::on_unpublish()
|
|||
destroy_forwarders();
|
||||
|
||||
encoder->on_unpublish();
|
||||
rtp->on_unpublish();
|
||||
rtc->on_unpublish();
|
||||
hls->on_unpublish();
|
||||
dash->on_unpublish();
|
||||
dvr->on_unpublish();
|
||||
|
|
|
|||
|
|
@ -54,7 +54,7 @@ class SrsNgExec;
|
|||
class SrsConnection;
|
||||
class SrsMessageHeader;
|
||||
class SrsHls;
|
||||
class SrsRtp;
|
||||
class SrsRtc;
|
||||
class SrsDvr;
|
||||
class SrsDash;
|
||||
class SrsEncoder;
|
||||
|
|
@ -360,8 +360,8 @@ private:
|
|||
private:
|
||||
// The format, codec information.
|
||||
SrsRtmpFormat* format;
|
||||
// rtp handler
|
||||
SrsRtp* rtp;
|
||||
// rtc handler
|
||||
SrsRtc* rtc;
|
||||
// hls handler.
|
||||
SrsHls* hls;
|
||||
// The DASH encoder.
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue