App Crash after updating WebRTC.xcframework WebRTC stack to m108

[Tuan Anh]

Hi Ant Media team,

I'm writing an application to stream 1 mp3 file to Ant Media Server.

I using these settings

          self.rtcAudioSession.lockForConfiguration()

        

        self.rtcAudioSession.useManualAudio = true

        self.rtcAudioSession.isAudioEnabled = false

        

        self.rtcAudioSession.unlockForConfiguration()

                   

         self.publisherClient =  AntMediaClient.init();

       self.publisherClient?.delegate = self

       self.publisherClient?.setOptions(url: self.clientUrl, streamId: self.publisherStreamId, token: "", mode: AntMediaClientMode.publish, enableDataChannel: false, captureScreenEnabled: true)

                

                //Tat Camera

     self.publisherClient?.setVideoEnable(enable: false);

     self.publisherClient?.setExternalVideoCapture(externalVideoCapture: true);

     self.publisherClient?.setTargetResolution(width: 1280, height: 720);

                //self.publisherClient?.setLocalView(container: self.localView, mode: .scaleAspectFit)

     self.publisherClient?.setExternalAudio(externalAudioEnabled: true)

     self.publisherClient?.initPeerConnection()

     self.publisherClient?.start()

 

Everything is OK with Old WebRTC (stack to m106) but it crashed after updating WebRTC (stack to m108)

It crashed at this function WebRTC`-[RTCVideoEncoderSettings maxFramerate]

Hope that you can help me find out the reason of this error. Thank you!

 

WebRTC`-[RTCVideoEncoderSettings maxFramerate]:

->  0x1026549c0 <+0>: ldr    w0, [x0, #0x18]

    0x1026549c4 <+4>: ret    

WebRTC`webrtc::ios_adm::AudioDeviceIOS::StartRecording:

    0x10275a5dc <+0>:   sub    sp, sp, #0x60

    0x10275a5e0 <+4>:   stp    x24, x23, [sp, #0x20]

    0x10275a5e4 <+8>:   stp    x22, x21, [sp, #0x30]

    0x10275a5e8 <+12>:  stp    x20, x19, [sp, #0x40]

    0x10275a5ec <+16>:  stp    x29, x30, [sp, #0x50]

    0x10275a5f0 <+20>:  add    x29, sp, #0x50

    0x10275a5f4 <+24>:  bl     0x10275d560               ; OUTLINED_FUNCTION_19

    0x10275a5f8 <+28>:  adrp   x23, 1253

    0x10275a5fc <+32>:  add    x23, x23, #0xdf4          ; "StartRecording"

    0x10275a600 <+36>:  adrp   x21, 1254

    0x10275a604 <+40>:  add    x21, x21, #0x644          ; "../../sdk/objc/native/src/audio/audio_device_ios.mm"

    0x10275a608 <+44>:  tbnz   w0, #0x0, 0x10275a62c     ; <+80>

    0x10275a60c <+48>:  adrp   x8, 1254

    0x10275a610 <+52>:  add    x8, x8, #0x678            ; "AudioDeviceIOS::"

    0x10275a614 <+56>:  stp    x8, x23, [sp, #0x10]

    0x10275a618 <+60>:  mov    w8, #0x891

    0x10275a61c <+64>:  stp    x21, x8, [sp]

    0x10275a620 <+68>:  adrp   x0, 1006

    0x10275a624 <+72>:  add    x0, x0, #0xb63            ; void rtc::webrtc_logging_impl::LogStreamer<>::Call<rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)13, rtc::webrtc_logging_impl::LogMetadata>, rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)9, char const*>, rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)9, char const*> >(rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)13, rtc::webrtc_logging_impl::LogMetadata> const&, rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)9, char const*> const&, rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)9, char const*> const&)::t

    0x10275a628 <+76>:  bl     0x102695ba8               ; rtc::webrtc_logging_impl::Log(rtc::webrtc_logging_impl::LogArgType const*, ...)

    0x10275a62c <+80>:  ldr    x0, [x19, #0xb8]

    0x10275a630 <+84>:  cbz    x0, 0x10275a638           ; <+92>

    0x10275a634 <+88>:  bl     0x10274efac               ; webrtc::FineAudioBuffer::ResetRecord()

    0x10275a638 <+92>:  add    x8, x19, #0xdc

    0x10275a63c <+96>:  ldar   w8, [x8]

    0x10275a640 <+100>: cbnz   w8, 0x10275a720           ; <+324>

    0x10275a644 <+104>: bl     0x10275d608               ; OUTLINED_FUNCTION_45

->  0x10275a648 <+108>: cmp    w0, #0x2

    0x10275a64c <+112>: b.ne   0x10275a720               ; <+324>

    0x10275a650 <+116>: ldr    x0, [x19, #0xb0]

    0x10275a654 <+120>: bl     0x10275f9ac               ; webrtc::ios_adm::VoiceProcessingAudioUnit::Start()

    0x10275a658 <+124>: cbz    w0, 0x10275a6f8           ; <+284>

    0x10275a65c <+128>: mov    x20, x0

    0x10275a660 <+132>: adrp   x8, 1421

    0x10275a664 <+136>: ldr    x0, [x8, #0xc8]

    0x10275a668 <+140>: adrp   x8, 1420

    0x10275a66c <+144>: ldr    x1, [x8, #0x7a8]

    0x10275a670 <+148>: bl     0x102b460a0               ; symbol stub for: objc_msgSend

    0x10275a674 <+152>: mov    x29, x29

    0x10275a678 <+156>: bl     0x102b460d0               ; symbol stub for: objc_retainAutoreleasedReturnValue

    0x10275a67c <+160>: mov    x19, x0

    0x10275a680 <+164>: adrp   x8, 1420

    0x10275a684 <+168>: ldr    x1, [x8, #0xbf8]

    0x10275a688 <+172>: mov    x2, x20

    0x10275a68c <+176>: bl     0x102b460a0               ; symbol stub for: objc_msgSend

    0x10275a690 <+180>: adrp   x8, 1420

    0x10275a694 <+184>: ldr    x22, [x8, #0xde8]

    0x10275a698 <+188>: adrp   x0, 1254

    0x10275a69c <+192>: add    x0, x0, #0x644            ; "../../sdk/objc/native/src/audio/audio_device_ios.mm"

    0x10275a6a0 <+196>: bl     0x102654410               ; RTCFileName

    0x10275a6a4 <+200>: mov    x29, x29

    0x10275a6a8 <+204>: bl     0x102b460d0               ; symbol stub for: objc_retainAutoreleasedReturnValue

    0x10275a6ac <+208>: mov    x21, x0

    0x10275a6b0 <+212>: adrp   x8, 1419

    0x10275a6b4 <+216>: ldr    x1, [x8, #0x608]

    0x10275a6b8 <+220>: stp    x23, x20, [sp, #0x10]

    0x10275a6bc <+224>: mov    w8, #0x11f

    0x10275a6c0 <+228>: stp    x0, x8, [sp]

    0x10275a6c4 <+232>: adrp   x2, 1423

    0x10275a6c8 <+236>: add    x2, x2, #0xe78            ; @"(%@:%d %s): StartRecording failed to start audio unit, reason %d"

    0x10275a6cc <+240>: mov    x0, x22

    0x10275a6d0 <+244>: bl     0x102b460a0               ; symbol stub for: objc_msgSend

    0x10275a6d4 <+248>: mov    x29, x29

    0x10275a6d8 <+252>: bl     0x102b460d0               ; symbol stub for: objc_retainAutoreleasedReturnValue

    0x10275a6dc <+256>: mov    x20, x0

    0x10275a6e0 <+260>: bl     0x102656174               ; OUTLINED_FUNCTION_1

    0x10275a6e4 <+264>: bl     0x10275d5e4               ; OUTLINED_FUNCTION_39

    0x10275a6e8 <+268>: bl     0x102654938               ; OUTLINED_FUNCTION_1

    0x10275a6ec <+272>: bl     0x102654940               ; OUTLINED_FUNCTION_2

    0x10275a6f0 <+276>: mov    w0, #-0x1

    0x10275a6f4 <+280>: b      0x10275a730               ; <+340>

    0x10275a6f8 <+284>: bl     0x102575f20               ; bool rtc::LogMessage::IsNoop<(rtc::LoggingSeverity)1>()

    0x10275a6fc <+288>: tbnz   w0, #0x0, 0x10275a720     ; <+324>

    0x10275a700 <+292>: adrp   x8, 1254

    0x10275a704 <+296>: add    x9, x8, #0x741            ; "Voice-Processing I/O audio unit is now started"

    0x10275a708 <+300>: mov    w8, #0x911

    0x10275a70c <+304>: stp    x8, x9, [sp, #0x8]

    0x10275a710 <+308>: str    x21, [sp]

    0x10275a714 <+312>: adrp   x0, 1006

    0x10275a718 <+316>: add    x0, x0, #0xb60            ; void rtc::webrtc_logging_impl::LogStreamer<>::Call<rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)13, rtc::webrtc_logging_impl::LogMetadata>, rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)9, char const*> >(rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)13, rtc::webrtc_logging_impl::LogMetadata> const&, rtc::webrtc_logging_impl::Val<(rtc::webrtc_logging_impl::LogArgType)9, char const*> const&)::t

    0x10275a71c <+320>: bl     0x102695ba8               ; rtc::webrtc_logging_impl::Log(rtc::webrtc_logging_impl::LogArgType const*, ...)

    0x10275a720 <+324>: mov    w0, #0x0

    0x10275a724 <+328>: add    x8, x19, #0xd8

    0x10275a728 <+332>: mov    w9, #0x1

    0x10275a72c <+336>: stlr   w9, [x8]

    0x10275a730 <+340>: ldp    x29, x30, [sp, #0x50]

    0x10275a734 <+344>: b      0x102651a7c               ; OUTLINED_FUNCTION_21

    0x10275a738 <+348>: mov    x1, x20

    0x10275a73c <+352>: mov    x20, x0

    0x10275a740 <+356>: mov    x0, x1

    0x10275a744 <+360>: b      0x10275a74c               ; <+368>

    0x10275a748 <+364>: bl     0x1026676c0               ; OUTLINED_FUNCTION_93

    0x10275a74c <+368>: bl     0x102b46094               ; symbol stub for: objc_release

    0x10275a750 <+372>: b      0x10275a75c               ; <+384>

    0x10275a754 <+376>: b      0x10275a758               ; <+380>

    0x10275a758 <+380>: mov    x20, x0

    0x10275a75c <+384>: bl     0x102654940               ; OUTLINED_FUNCTION_2

    0x10275a760 <+388>: bl     0x10264ecbc               ; OUTLINED_FUNCTION_18

 

[Tuan Anh]

if I delete this line         self.rtcAudioSession.useManualAudio = true

It doesn't crash anymore. However, the audio sent to Server if from the device microphone, not mp3 file anymore. So that I want to still use this line self.rtcAudioSession.useManualAudio = true

[Selim]

Hi @Tuan Anh,

Thank you for reporting this issue. I'm little bit confused. Which part did you add your mp3 file to?

Best Regards,

Selim

[Tuan Anh]

Hi @Selim,

To stream Mp3 file, I get Audio Data from Mp3 File  and sent these data through this function self.publisherClient?.deliverExternalAudio(sampleBuffer: cmSampleBuffer!)

Everything works OK with old WebRTC version (m96) . I received audio data on Server and can Listened sound on Web Preview.  But now with new WebRTC, it crashed.

Best regards,

Tuấn Anh

 

 

Edited January 17 by Tuan Anh

[Selim]

Hi @Tuan Anh,

Have a good day. It seems that this issue will take some time. Could you please add an issue to this link: https://github.com/ant-media/Ant-Media-Server/issues We are following development requirements on this link.

Best Regards,

Selim