open-webrtc-toolkit / owt-deps-webrtc Goto Github PK

how can we judge this packet to be the first packet of a frame?
it is just first packet of a NAL

Same problem with h265

As above, how to success build and compile 70-sdk, can someone guide me?

Instead of #ifndef DISABLE_H265, it might be more convenient to write #if defined(RTC_ENABLE_H265)

I've looked at branches 76, 83, and 88 and they all have this around line 1279:

bool ParseExtmap(const std::string& line,
                 RtpExtension* extmap,
                 SdpParseError* error) {
  // RFC 5285
  // a=extmap:<value>["/"<direction>] <URI> <extensionattributes>
  std::vector<std::string> fields;
  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpaceChar, &fields);
  const size_t expected_min_fields = 2;
  if (fields.size() < expected_min_fields) {
    return ParseFailedExpectMinFieldNum(line, expected_min_fields, error);
  }

But now (e.g. > 71) there is RFC 8285 being used by Chrome on Droid and windows, which supports a=extmap-allow-mixed to be used with no value. I can't explain why I am just now seeing this on some devices, but from what can gather there is a version span of Chrome up to version 71 that will strip this out of the SDP, anything greater will keep it in, and then this line will fail because it expects 2 fields.

Can someone enlighten me on if this code should change or is it expected that the caller (i.e. chrome) should strip this out. For now I am going to have to work out how to properly allow this to parse to get our stuff working again. Thanks in advance for any feedback.

(packet_buffer.cc:243): PacketBuffer is already at max size (2048), failed to increase size.
(packet_buffer.cc:108): Clear PacketBuffer and request key frame.
(packet_buffer.cc:243): PacketBuffer is already at max size (2048), failed to increase size.
(packet_buffer.cc:108): Clear PacketBuffer and request key frame.
(packet_buffer.cc:243): PacketBuffer is already at max size (2048), failed to increase size.
(packet_buffer.cc:108): Clear PacketBuffer and request key frame.

When will it be merged and updated to M98?

when " build libwebrtc for OWT Android SDK with scripts/build_android.py " error,
无法生存文件obj/third_party/webrtc/rtc_base/logging/logging.o

building libjingle_peerconnection_so for arm release ninja: Entering directory/home/hou/chromium/src/out/releasearm'
[6/929] CXX obj/third_party/webrtc/rtc_base/logging/logging.o
FAILED: obj/third_party/webrtc/rtc_base/logging/logging.o 错误原因是：../../third_party/webrtc/rtc_base/logging.cc:53:36: error: use of undeclared identifier 'LS_NONE'
static LoggingSeverity g_min_sev = LS_NONE;
^
../../third_party/webrtc/rtc_base/logging.cc:54:36: error: use of undeclared identifier 'LS_NONE'
static LoggingSeverity g_dbg_sev = LS_NONE;
^
../../third_party/webrtc/rtc_base/logging.cc:82:6: error: use of undeclared identifier 'LogSink'
void LogSink::OnLogMessage(const std::string& msg,

......
`

fatal error: too many errors emitted, stopping now [-ferror-limit=] 20 errors generated. [11/929] CXX obj/third_party/webrtc/pc/rtc_pc_base/srtpsession.o ninja: build stopped: subcommand failed.

the code have h265 rtp packet operation,but not have h265 encoder and decoder in modules/video_coding/codecs/h265

Following items are needed for low latency mode.

• Realtime mode in pacer
• Start bitrate
• Opus channels
• Priority of rtp_send_controller task queue
• Bypass audio processing
• Remove GPRA

Hi, using 83-sdk for android, HW H264 encoder will hang after several seconds on Huawei P30(kirin 980).
E org.webrtc.Logging: HardwareVideoEncoder: Dropped frame, encoder queue full
E org.webrtc.Logging: HardwareVideoEncoder: Dropped frame, encoder queue full
...
Would you give some suggestion?

such as h265_encoder_xxx.cc

Ubuntu 18.04
MCU Server V4.3

client report 'ice close' , and webrtc agent report log :

2019-11-29 16:31:47,220 - WARN: LibNiceConnection - id: 759289002149473500, message: failed and not received all candidates, newComponentState:4
2019-11-29 16:31:47,576 - WARN: LibNiceConnection - id: 758165339333195300, message: failed and not received all candidates, newComponentState:4
2019-11-29 16:31:58,479 - WARN: LibNiceConnection - id: 115807156049238420, message: failed and not received all candidates, newComponentState:4
2019-11-29 16:31:59,927 - WARN: LibNiceConnection - id: 468377342159567800, message: failed and not received all candidates, newComponentState:4
2019-11-29 16:32:05.607 - WARN: WorkingNode - Exiting on SIGTERM

原因是 h265进入后台会重置解码器，但是对方并没有发送关键帧过来，导致解码p帧失败。

I am trying to build the owt webrtc on a Win10, Visual Studio 2019 machine.

However upon trying to generate the build files via:
C:\Git\owt_webrtc\src>gn gen out\Debug --args="is_debug=true rtc_use_h264=false target_os=\"win\" target_cpu=\"x64\"
I get the following error:

C:\Git\owt_webrtc\src>gn gen out\Debug --args="is_debug=true rtc_use_h264=false target_os=\"win\" target_cpu=\"x64\"
ERROR at //webrtc.gni:15:1: Can't load input file.
import("//build_overrides/ssl/ssl.gni")
^-------------------------------------
Unable to load:
  C:/Git/owt_webrtc/src/build_overrides/ssl/ssl.gni
I also checked in the secondary tree for:
  C:/Git/owt_webrtc/src/build/secondary/build_overrides/ssl/ssl.gni
See //BUILD.gn:17:1: whence it was imported.
import("webrtc.gni")

And its true there are no files there:

Does anybody know why the files are missing ?

on honor v20 device, this line will freeze the video capture.

Reproduce in owt-client-android conference sample:

1. use Camera2Capturer. (do some modify in OwtVideoCapture)
2. OwtVideoCapturer.create use 3840x2160 resolution and set isCameraFront to false.(front camera not support 4k)
3. start the sample, then the video capture area draw nothing.
4. remove Camera2Session CONTROL_VIDEO_STABILIZATION_MODE_ON, restart the sample, everything going fine.

Compilation error reported on Linux for branch cloudgaming-83:

../../third_party/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc:79:7: error: use of undeclared identifier 'sscanf_s'
      sscanf_s(expr_str.c_str(), "Enabled-%u,%u,%u", start_bitrate_kbps,
      ^

how can i fix it？

1. i just git clone source on my pc computer
   2 i don't know how i can use webrtc_official commond to gclient sync other source code.
2. or can you tell me what version of webrtc i should download on my computer, thanks !

ERROR at //build/config/android/internal_rules.gni:162:21: Can't load input file.
deps += [ "$_target_label$build_config_target_suffix" ]

When can AV1 confluence be supported？

Looking at the RtpPacketizerH265::GeneratePackets function, you can see code like this:

if (fragment_len> single_packet_capacity)
     {
       PacketizeFu(i);
       ++i;
     }
     else
     {
       Packetize Single Nalu(i);
       ++i;
     }

I think it's better to use PacketizeAp instead of PacketizeSingleNalu here. Is this intended? Or is it a mistake? I wonder why the H265Packetizer uses SingleNalu instead of STAB-A if it is intended to waste the remaining space of a RTP packet.