Send/Receive FIR Packet

[Luke Mauldin]

All,

I am attempting to send an FIR packet to a Chrome browser using the native C++ WebRTC libraries and I am using the code as rtcp_sender.cc.  I know that the packet is going from the C++ application to the browser because I have stepped through it in debug mode, however it seems like the browser is ignoring the packet.  I am running chrome with the following settings: ./chrome --enable-logging --vmodule=*libjingle/source/talk/*=3  

However, I am not able to determine why Chrome is ignoring the FIR packet.  Can anyone please help me out?

Luke

[Adam Fineberg]

You might want to make sure the correct SSRC is being sent.
Additionally, check in the log (you'll need a debug build with webrtc
trace enabled) for the message:

SIG [RTCP] Incoming PLI from SSRC:0x%x

This message should be printed for received FIRs as well as PLIs if the
SSRC in rtcpPacket.FIRItem.SSRC matches a known stream SSRC.

Regards,
Adam

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[Luke Mauldin]

Adam, I have a built my own version of debug chromium. How do I turn on webrtc trace?

Also, it looks like there are two SSRC fields in the data packet. Which one should be set?

Luke

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[Adam Fineberg]

You need to set enable_trace = 1 in the gyp file.  The SSRC should be the FIRItem.SSRC.

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Regards,
Adam

[Luke Mauldin]

Adam,

Which gyp file should I update?  I did the following search in the chromium source tree and didn't find any matching files:

grep -r --include="*.gyp" -i "enable_trace"

On Tuesday, October 29, 2013 4:32:28 PM UTC-5, Adam Fineberg wrote:

You need to set enable_trace = 1 in the gyp file.  The SSRC should be the FIRItem.SSRC.

On 10/29/13 2:09 PM, Luke Mauldin wrote:

Adam, I have a built my own version of debug chromium. How do I turn on webrtc trace?

Also, it looks like there are two SSRC fields in the data packet. Which one should be set?

Luke

On Tuesday, October 29, 2013, Adam Fineberg wrote:

You might want to make sure the correct SSRC is being sent. Additionally, check in the log (you'll need a debug build with webrtc trace enabled) for the message:

SIG [RTCP] Incoming PLI from SSRC:0x%x

This message should be printed for received FIRs as well as PLIs if the SSRC in rtcpPacket.FIRItem.SSRC matches a known stream SSRC.

Regards,
Adam

On 10/29/13 11:22 AM, Luke Mauldin wrote:

All,

I am attempting to send an FIR packet to a Chrome browser using the native C++ WebRTC libraries and I am using the code as rtcp_sender.cc.  I know that the packet is going from the C++ application to the browser because I have stepped through it in debug mode, however it seems like the browser is ignoring the packet.  I am running chrome with the following settings: ./chrome --enable-logging --vmodule=*libjingle/source/talk/*=3
However, I am not able to determine why Chrome is ignoring the FIR packet.  Can anyone please help me out?

Luke
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Regards,
Adam

[Adam Fineberg]

third_party/webrtc/build/common.gypi, look for enable_tracing

[Luke Mauldin]

I modified the common.gypi file mentioned above and did a full rebuild of Chromium.  Now when I launch Chromium I get some additional logging, but I do not see any messages like you described above saying "Incoming PLI" or "Incoming FIR" packet.  I am attempting to send both FLI and FIR packets using the methods below.  It is like Chromium is either not receiving the packets or not accepting them?  Is there any additional logging I can enable or where should I set breakpoints for debugging?

std::shared_ptr<talk_base::Buffer> BuildPLIPacket(uint32 localSsrc, uint32 remoteSsrc)

{

  const int bufferSize = 20;

  const int bufferCapacity = bufferSize + 14;

  uint8_t rtcpbuffer[bufferSize];

  std::fill(std::begin(rtcpbuffer), std::end(rtcpbuffer), 0);

  int pos = 0;

   

    // add picture loss indicator

    uint8_t FMT = 1;

    rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;

    rtcpbuffer[pos++]=(uint8_t)206;

    //Used fixed length of 2

    rtcpbuffer[pos++]=(uint8_t)0;

    rtcpbuffer[pos++]=(uint8_t)(2);

    // Add our own SSRC

    AssignUWord32ToBuffer(rtcpbuffer+pos, localSsrc);

    pos += 4;

    // Add the remote SSRC

    AssignUWord32ToBuffer(rtcpbuffer+pos, remoteSsrc);

    pos += 4;

  

  //Build and send buffer packet

  std::cout << "Built PLI packet for localSsrc: " << localSsrc

            << " -- remoteSsrc: " << remoteSsrc << " -- pos: " << pos

            << std::endl;

  return std::make_shared<talk_base::Buffer>(rtcpbuffer, bufferSize,

                                             bufferCapacity);

}

static int firSequenceNumber = 0;

std::shared_ptr<talk_base::Buffer> BuildFIRPacket(uint32 localSsrc,

                                                  uint32 remoteSsrc) {

  //Build FIR packet

  const int bufferSize = 20;

  const int bufferCapacity = bufferSize + 14;

  uint8_t rtcpbuffer[bufferSize];

  std::fill(std::begin(rtcpbuffer), std::end(rtcpbuffer), 0);

  int pos = 0;

  // add full intra request indicator

  uint8_t FMT = 4;

  rtcpbuffer[pos++] = (uint8_t) 0x80 + FMT;

  rtcpbuffer[pos++] = (uint8_t) 206;

  //Length of 4

  rtcpbuffer[pos++] = (uint8_t) 0;

  rtcpbuffer[pos++] = (uint8_t)(4);

  // Add our own SSRC

  AssignUWord32ToBuffer(rtcpbuffer + pos, localSsrc);

  pos += 4;

  // RFC 5104     4.3.1.2.  Semantics

  // SSRC of media source

  rtcpbuffer[pos++] = (uint8_t) 0;

  rtcpbuffer[pos++] = (uint8_t) 0;

  rtcpbuffer[pos++] = (uint8_t) 0;

  rtcpbuffer[pos++] = (uint8_t) 0;

  // Additional Feedback Control Information (FCI)

  AssignUWord32ToBuffer(rtcpbuffer + pos, remoteSsrc);

  pos += 4;

  rtcpbuffer[pos++] = (uint8_t)(firSequenceNumber);

  rtcpbuffer[pos++] = (uint8_t) 0;

  rtcpbuffer[pos++] = (uint8_t) 0;

  rtcpbuffer[pos++] = (uint8_t) 0;

  firSequenceNumber++;

  //Build and send buffer packet

  std::cout << "Built FIR packet for localSsrc: " << localSsrc

            << " -- remoteSsrc: " << remoteSsrc << " -- pos: " << pos

            << std::endl;

  return std::make_shared<talk_base::Buffer>(rtcpbuffer, bufferSize,

                                             bufferCapacity);

}

//Sending code - I have also tried it with BuildPLIPacket

firPacket = BuildFIRPacket(presenterVideoChannel->GetSendSsrc(),

                                 presenterVideoChannel->GetRecvSsrc());

      presenterVideoChannel->SendRtcp(firPacket.get());

[Adam Fineberg]

Set your breakpoint in webrtc/modules/rtp_rtcp_source/rtcp_receiver.cc HandleFIR and/or HandlePLI and you should see if the packets are being received and why they are dropped (if they are).

[Sergio Garcia Murillo]

It would be great if anyone of you could share the binary built of chromium with webrtc traces enabled (especially for windows). It would be really helpful for quite a lot of us.. ;)

Best regards
Sergio

[Lorenzo Miniero]

+1!

Lorenzo

[Kaiduan Xie]

You can build Chromium with webrtc trace,

On Mac run Chrome as below,

Chrome --enable-logging > log 2>&1

/Kaiduan

The following lists the changes, the changes are marked in pink.

1) Libjingle,

third_party/libjingle/overrides/talk/base/logging.cc

$ svn diff .

Index: overrides/talk/base/logging.cc

===================================================================

--- overrides/talk/base/logging.cc (revision 221536)

+++ overrides/talk/base/logging.cc (working copy)

@@ -151,7 +151,7 @@

 DiagnosticLogMessage::~DiagnosticLogMessage() {

   print_stream_ << extra_;

   const std::string& str = print_stream_.str();

-  if (log_to_chrome_)

+  //if (log_to_chrome_)

     LOG_LAZY_STREAM_DIRECT(file_name_, line_, severity_) << str;

   if (g_logging_delegate_function && severity_ <= LS_INFO) {

     print_stream_with_timestamp_ << str;

@@ -290,6 +290,7 @@

kxie@ci0000003749415 ~/chromium/src/third_party/libjingle/source/talk/media/webrtc $ svn diff .

Index: webrtcvoiceengine.cc

===================================================================

--- webrtcvoiceengine.cc (revision 4650)

+++ webrtcvoiceengine.cc (working copy)

@@ -1070,8 +1070,9 @@

 }

 

 void WebRtcVoiceEngine::SetTraceFilter(int filter) {

-  log_filter_ = filter;

-  tracing_->SetTraceFilter(filter);

+  //log_filter_ = filter;

+  log_filter_ = webrtc::kTraceAll;

+  tracing_->SetTraceFilter(log_filter_);

 }

 

 // We suppport three different logging settings for VoiceEngine:

@@ -1159,9 +1160,10 @@

     LOG_V(sev) << msg;

   } else {

     std::string msg(trace + 71, length - 72);

-    if (!ShouldIgnoreTrace(msg)) {

-      LOG_V(sev) << "webrtc: " << msg;

-    }

+    //if (!ShouldIgnoreTrace(msg)) {

+    LOG(LS_INFO) << "webrtc: " << msg;

+    //  LOG_V(sev) << "webrtc: " << msg;

+    //}

   }

 }

 

Index: webrtcvideoengine.cc

===================================================================

--- webrtcvideoengine.cc (revision 4650)

+++ webrtcvideoengine.cc (working copy)

@@ -1324,10 +1324,11 @@

     LOG_V(sev) << msg;

   } else {

     std::string msg(trace + 71, length - 72);

-    if (!ShouldIgnoreTrace(msg) &&

-        (!voice_engine_ || !voice_engine_->ShouldIgnoreTrace(msg))) {

-      LOG_V(sev) << "webrtc: " << msg;

-    }

+    //if (!ShouldIgnoreTrace(msg) &&

+    //    (!voice_engine_ || !voice_engine_->ShouldIgnoreTrace(msg))) {

+    //  LOG_V(sev) << "webrtc: " << msg;

+    LOG(LS_INFO) << "webrtc: " << msg;

+    //}

   }

 }

2) webrtc

kxie@ci0000003749415 ~/chromium/src/third_party/webrtc/system_wrappers $ svn diff .

Index: source/trace_impl.cc

===================================================================

--- source/trace_impl.cc (revision 4650)

+++ source/trace_impl.cc (working copy)

@@ -29,13 +29,16 @@

 #pragma warning(disable:4355)

 #endif  // _WIN32

 

+#define WEBRTC_DIRECT_TRACE

+

 namespace webrtc {

 

 const int Trace::kBoilerplateLength = 71;

 const int Trace::kTimestampPosition = 13;

 const int Trace::kTimestampLength = 12;

 

-static uint32_t level_filter = kTraceDefault;

+static uint32_t level_filter = kTraceAll;

+//static uint32_t level_filter = kTraceDefault;

 

 // Construct On First Use idiom. Avoids "static initialization order fiasco".

 TraceImpl* TraceImpl::StaticInstance(CountOperation count_operation,

@@ -43,11 +46,13 @@

   // Sanities to avoid taking lock unless absolutely necessary (for

   // performance reasons). count_operation == kAddRefNoCreate implies that a

   // message will be written to file.

+#if 0 // !xkd!

   if ((level != kTraceAll) && (count_operation == kAddRefNoCreate)) {

     if (!(level & level_filter)) {

       return NULL;

     }

   }

+#endif

   TraceImpl* impl =

     GetStaticInstance<TraceImpl>(count_operation);

   return impl;

[Bo Xu]

could you tell me where(which cc file) I can run the following part of your code to send out FIR packet?

...

//Sending code - I have also tried it with BuildPLIPacket

firPacket = BuildFIRPacket(presenterVideoChannel->GetSendSsrc(),presenterVideoChannel->GetRecvSsrc());

presenterVideoChannel->SendRtcp(firPacket.get());

...

Thanks!

Bo Xu

[Luke Mauldin]

Sorry, I don't have access to that code anymore.

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Regards,
Adam

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Regards,
Adam

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