LEB(low-latency live streaming)
Supports audio and video streaming, supports separate audio streaming and video streaming, supports H.264, H.265, AAC, Opus, and other codecs, supports AAC/LATM format.
``
typedef struct LebConfig {
// default set 0, and obs has supported.
int is_whip;
// push stream url:
// "webrtc://webrtcpush.myqcloud.com/live/xxx"
const char* stream_url;
// signal ip or domation，default value as "webrtcpush.myqcloud.com"
const char* signal_address;
//if push video stream, please set 1. or 0
int enable_video;
// if push audio stream, please set 1. or 0
int enable_audio;
// Statistics callback interval, internal default 5 second. sorry this valueable has ms suffix.
float stats_period_ms;
//Maximum number of tasks for scheduling one thread. default 100.
uint32_t max_tasks_per_thread;
//to-do
//Set by sdk user. When a crash occurs, it will be set at the next startup.
// [
// {
// "time:":"2023-03-23 20:43:52",
// "sdkver":"1.0.1",
// "os":"15.5",
// "device" :"linux",
// "abi" : "x64",
// "stack" "crash stack",
// },
// {
// "time:":"2023-03-23 20:43:52",
// "sdkver":"1.0.1",
// "os":"15.5",
// "device" :"ios",
// "abi" : "arm64",
// "stack" "crash stack",
// }
// ]
const char* last_bug_json;
//to-do
const char* report_api;
//TO-DO
} LebConfig;
//Initialize the configuration
LebConfig config;
config.is_whip = 0;
config.stream_url = pushurl;
config.signal_address = "webrtcpush.myqcloud.com";//"webrtcpush.myqcloud.com"
config.enable_video = 1;
config.enable_audio = 1;
config.stats_period_ms = 0.f;
config.max_tasks_per_thread = 200;
config.last_bug_json = NULL;
config.report_api = NULL;
//create a leb pusher, and get pusher handler(LebPusherHandle).
// this struct is seted by sdk.
typedef struct LebPusherHandle {
void* context;
void* internal_handle;
LebConfig config;
LebCallback callback;
// caller set log callback
void (*onLogInfo)(void* context,
const char* tag,
LebLogLevel,
const char* message);
//caller set stat callback
void (*onStatsInfo)(void* context, LebStats stats);
//caller set error callback
void (*onError)(void* context, LebErrorCode error);
//caller set error callback
void (*onErrorInfo)(void* context, LebErrorCode error, const char* strerror);
//caller set connection state callback
void (*onConnectionChange)(void* context, LebNetState state);
} LebPusherHandle;
typedef struct LebCallback {
// caller use this function pointer to set video stream metadata.
OnVideoInfo onVideoInfo;
// caller use this function pointer to set audio stream metadata.
OnAudioInfo onAudioInfo;
// caller use this function pointer to write into video media data.
OnEncodedVideo onEncodedVideo;
// caller use this function pointer to write into audio media data.
OnEncodedAudio onEncodedAudio;
// // caller use this function pointer to write into stream metadata.
OnMetaData onMetaData;
//runtime control function pointer
OnCtxControl onCtxControl;
} LebCallback;
LebPusherHandle *handle_ = OpenLebPusherV2(this, kInfo, config);
//register callback function pointer
a>connection state callback
handle_->onConnectionChange = OnConnectionChangeCallback;
b> error callback
handle_->onError = OnErrorCallbck;
c> log callback
handle_->onLogInfo = OnLogCallback;
d>statistic callback
handle_->onStatsInfo = OnStatsCallback;
//set leb push stream video/audio meta data.
handle_->callback.onVideoInfo(handle_, video_info_);
handle_->callback.onAudioInfo(handle_, audio_info_);
StartLebPusher(handle_, config);
LebEncodedVideoFrame videoframe;
handle_->callback.onEncodedVideo(handle_, videoframe);
LebEncodedAudioFrame audioframe;
handle_->callback.onEncodedAudio(handle_, audioframe);
StopLebPusher(handle_);
CloseLebPusher(handle_);
graph TD;
A[initialize LebConfig]-->B[create Pusher:OpenLebPusherV2];
B-->C[start Pusher:StartLebPusher];
C-->D[write into A/V media data onEncodedVideo/onEncodedAudio];
D-->StopLebPusher;
StopLebPusher-->CloseLebPusher;
Refer to the interface description above，Or the demo code provided in the sdk.
Integrate webrtc streaming function through ffmpeg. Directly use ffmpeg to implement webrtc leb Push streaming。
You can refer to the webrtc_muxer.c file in the sdk。if you want to get the muxer code, please in touch with developer
AVOutputFormat ff_webrtc_muxer = {
.name = "webrtc",
.long_name = NULL_IF_CONFIG_SMALL("webrtc"),
.mime_type = "video/webrtc",
.extensions = "webrtc",
.priv_data_size = sizeof(LibWebRTCContext),
.audio_codec = AV_CODEC_ID_OPUS | AV_CODEC_ID_AAC,
.video_codec = AV_CODEC_ID_H264,
.init = webrtc_init,
.write_header = webrtc_write_header,
.write_packet = webrtc_write_packet,
.write_trailer = webrtc_write_trailer,
.deinit = webrtc_free,
.flags = AVFMT_GLOBALHEADER | AVFMT_ALLOW_FLUSH | AVFMT_TS_NEGATIVE | AVFMT_NOFILE,
.codec_tag = (const AVCodecTag* const []){ codec_webrtc_tags, 0 },
//.check_bitstream = webrtc_check_bitstream,
.priv_class = &webrtc_class,
};
ffmpeg muxer plugin params.
-webrtc_signal this specify the signaling server
-is_aac_latm support aac out of band.
-fillsei_avdiff add debug sei for computer audio - video synchronization timestamp difference
-is_whip support whip protocol
-b_frame support b frame push
-bitrate_cfg set bitrate in config
-stats_period_ms stat callback interval
-stat_api report api
-stream_pri stream priority
ffmpeg example:
ffmpeg -re -stream_loop -1 -i /Users/ts/test/machine.flv -c:v copy -acodec copy -bsf:a aac_adtstoasc -f webrtc "webrtc://xxx/live/webrtc_test_7007?txSecret=f9269a11eab5b80c8fe540c7b68a084b&txTime=65C105FF"
SDK donwload latest version：
https://media-frame-1258344699.cos.ap-nanjing.myqcloud.com/tstan/xp2p/leb_push_sdk_linux_release_v_1.1.2_34e32e4ebe_md96f396a453_2023_11_01_20_57.zip
sdk content：
leb_push_sdk
├── leb_push_api.h
└── libLebPusher_so.so
ffmpeg integrated with sdk:
sdk_demo/
└── ffmpeg_g
if you want to get the sdk, please in touch with developer https://github.com/troman123
TO-DO
//
if you want to get the sdk, please in touch with developer https://github.com/troman123
//