owt-server 数据流程

最新推荐文章于 2024-08-21 09:48:11 发布

古灵古怪

最新推荐文章于 2024-08-21 09:48:11 发布

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分类专栏： webrtc之OWT-Server 音视频编解码

本文链接：https://blog.youkuaiyun.com/dtryl/article/details/109031701

音视频编解码同时被 2 个专栏收录

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webrtc之OWT-Server

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本文档介绍了OpenWebRTC Toolkit（OWT）服务器如何使用RabbitMQ创建队列，并展示了其组件架构图。在运行过程中，遇到了音频混流的问题，错误源于WebRTC编译时的符号未定义。通过分析日志，定位到错误源为`WebRtcVad_InitCore`未找到，最终通过修改源代码解决了问题。同时，列出了OWT运行时连接的worker目的类型，包括录音、会议、音频等。

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1，/root/Projects/owt-server/dist/management_api/rpc/rpc.js

//Create the queue for send messages
clientQueue = connection.queue('', function (q) {
log.info('ClientQueue ' + q.name + ' is open'+',clientQueue.name:'+clientQueue.name);
clientQueue.bind('owtRpc', clientQueue.name);
clientQueue.subscribe(function (message) {
if (map[message.corrID] !== undefined) {
map[message.corrID].fn[message.type](message.data, message.err);
clearTimeout(map[message.corrID].to);
delete map[message.corrID];
}
});
});

发现owt采用的队列名是rabbitmq-server自己随机产生的队列名

2， owt-server组件架构图:

root@cluster_manager$node .
2020-10-13 10:23:55.786 - INFO: AmqpClient - Connecting to rabbitMQ server OK, options: [Object: null prototype] { host: 'localhost', port: 5672, login: 'vzan' }
2020-10-13 10:23:55.793 - INFO: Main - Cluster manager up! id: 403728697
2020-10-13 10:23:55.794 - INFO: ClusterManager - Run as candidate.
2020-10-13 10:23:55.994 - INFO: ClusterManager - Run as master.
2020-10-13 10:23:56.012 - INFO: ClusterManager - Cluster manager is in service as master!
2020-10-13 10:23:56.043 - WARN: ClusterManager - workerJoin, purpose: video |worker: video-8e2cd52dcc415797a799@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'video',
state: 2,
max_load: 0.85,
capacity:
{ isps: [],
regions: [],
video: { decode: [Array], encode: [Array] } } }
2020-10-13 10:23:56.065 - WARN: ClusterManager - workerJoin, purpose: webrtc |worker: webrtc-7c8b15c2b2a839c97aef@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'webrtc',
state: 2,
max_load: 0.85,
capacity: { isps: [], regions: [] } }
2020-10-13 10:23:56.176 - WARN: ClusterManager - workerJoin, purpose: sip |worker: sip-ee60010a8ea684540b55@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'sip',
state: 2,
max_load: 0.85,
capacity: {} }
2020-10-13 10:23:56.269 - WARN: ClusterManager - workerJoin, purpose: analytics |worker: analytics-e0a44bf8aac93aa51216@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'analytics',
state: 2,
max_load: 0.85,
capacity:
{ isps: [],
regions: [],
algorithms:
[ 'dc51138a8284436f873418a21ba8cfa7',
'b849f44bee074b08bf3e627f3fc927c7' ] } }
2020-10-13 10:23:56.281 - WARN: ClusterManager - workerJoin, purpose: portal |worker: portal-5f2c0647fa517132fa9b@192.168.60.113 info: { ip: '192.168.60.113',
hostname: '',
port: 8080,
via_host: '',
ssl: true,
state: 2,
max_load: 0.85,
capacity: { isps: [], regions: [] } }
2020-10-13 10:23:56.326 - WARN: ClusterManager - workerJoin, purpose: streaming |worker: streaming-ab011678ccb67fa465f8@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'streaming',
state: 2,
max_load: 0.85,
capacity: { isps: [], regions: [] } }
2020-10-13 10:23:56.428 - WARN: ClusterManager - workerJoin, purpose: audio |worker: audio-d0e383715b1bd505912a@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'audio',
state: 2,
max_load: 0.85,
capacity: { isps: [], regions: [] } }
2020-10-13 10:23:56.434 - WARN: ClusterManager - workerJoin, purpose: conference |worker: conference-7c3c3b015c97d1e742e5@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'conference',
state: 2,
max_load: 0.85,
capacity: {} }
2020-10-13 10:23:56.470 - WARN: ClusterManager - workerJoin, purpose: recording |worker: recording-f73079288a2b5013c919@192.168.60.113 info: { ip: '192.168.60.113',
purpose: 'recording',
state: 2,
max_load: 0.85,
capacity: {} }

单独运行cluster_manager之后,发现连接上来的purpose有:

recording
conference
audio
streaming
portal
analytics
sip
webrtc
video

4，遇到音频混流问题:

==> logs/audio-d0e383715b1bd505912a@192.168.60.113_1.log <==
node: symbol lookup error: /root/Projects/owt-server/dist/audio_agent/audioMixer/build/Release/audioMixer.node: undefined symbol: WebRtcVad_InitCore

==> logs/audio-agent.stdout <==
2020-10-13 10:31:42.705 - INFO: NodeManager - Node audio-d0e383715b1bd505912a@192.168.60.113_1 is closed on unexpected code: 127
2020-10-13 10:31:42.776 - INFO: NodeManager - Node audio-d0e383715b1bd505912a@192.168.60.113_2 is closed on unexpected code: 127

==> logs/conference-7c3c3b015c97d1e742e5@192.168.60.113_0.log <==
2020-10-13 10:31:43.561 - ERROR: RoomController - subscribe failed, reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_0.generate
2020-10-13 10:31:43.561 - INFO: Conference - roomController.subscribe failed, reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_0.generate
2020-10-13 10:31:43.561 - INFO: Conference - Exception: roomController.subscribe failed, reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_0.generate
2020-10-13 10:31:44.701 - ERROR: RoomController - spreadStream failed, stream_id: 606272553558733400 reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_1.publish
2020-10-13 10:31:44.701 - INFO: Conference - roomController.mix failed, reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_1.publish
2020-10-13 10:31:44.702 - WARN: Conference - failed in controlStream, reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_1.publish

==> logs/app.stdout <==
Tue, 13 Oct 2020 02:31:44 GMT express deprecated res.send(status): Use res.sendStatus(status) instead at samplertcservice.js:339:9
PATCH /rooms/5f8066b6b67d299660406f1b/streams/606272553558733400 500 2065.788 ms - 21

==> logs/conference-7c3c3b015c97d1e742e5@192.168.60.113_0.log <==
2020-10-13 10:31:44.766 - ERROR: RoomController - spreadStream failed, stream_id: 606272553558733400 reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_2.publish
2020-10-13 10:31:44.767 - INFO: Conference - roomController.mix failed, reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_2.publish
2020-10-13 10:31:44.767 - WARN: Conference - failed in controlStream, reason: Timeout to make rpc to audio-d0e383715b1bd505912a@192.168.60.113_2.publish

==> logs/app.stdout <==
PATCH /rooms/5f8066b6b67d299660406f1b/streams/606272553558733400 500 2154.875 ms - 21

出现这个问题的原因是编译webrtc的时候出错了,可是非常坑的是owt编译脚本并不停止下来并提示错误,误以为正常，直到运行的时候出现问题，又茫然不知所措。

../webrtc/common_audio/vad/vad_core.c:118:1: error: attributes should be specified before the declarator in a function definition
118 | static inline int32_t OverflowingMulS16ByS32ToS32(int16_t a, int32_t b)
| ^~~~~~
../webrtc/common_audio/vad/vad_core.c: In function ‘GmmProbability’:
../webrtc/common_audio/vad/vad_core.c:391:22: warning: implicit declaration of function ‘OverflowingMulS16ByS32ToS32’ [-Wimplicit-function-declaration]
391 | tmp2_s32 = OverflowingMulS16ByS32ToS32(tmp_s16, tmp1_s32);

修改代码吧，但是要注意控制脚本的运行，防止自动rm,重新git同步代码，不然白改了:

//文件:vad_core.cc

static inline int32_t OverflowingMulS16ByS32ToS32(int16_t a, int32_t b)
    RTC_NO_SANITIZE("signed-integer-overflow") {
  return a * b;
}

===>

static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow") 
OverflowingMulS16ByS32ToS32(int16_t a, int32_t b) {
  return a * b;
}



//entropy_coding.c

static inline int32_t OverflowingMulS16S32ToS32(int16_t a, int32_t b)
    RTC_NO_SANITIZE("signed-integer-overflow") {
  return a * b;
}
static inline int32_t OverflowingAddS32S32ToS32(int32_t a, int32_t b)
    RTC_NO_SANITIZE("signed-integer-overflow") {
  return a + b;
}
static inline int32_t OverflowingSubS32S32ToS32(int32_t a, int32_t b)
    RTC_NO_SANITIZE("signed-integer-overflow") {
  return a - b;
}



===>

static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
OverflowingMulS16S32ToS32(int16_t a, int32_t b) {
  return a * b;
}
static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
OverflowingAddS32S32ToS32(int32_t a, int32_t b) {
  return a + b;
}
static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")  
OverflowingSubS32S32ToS32(int32_t a, int32_t b) {
  return a - b;
}



//lattice.c



static inline int32_t OverflowingLShiftS32(int32_t x, int shift)
    RTC_NO_SANITIZE("shift") {
  return x << shift;
}

==>

static inline int32_t RTC_NO_SANITIZE("shift") 
OverflowingLShiftS32(int32_t x, int shift) {
  return x << shift;
}