anyrtcio-community / anyrtc-rtmp-opensource Goto Github PK

是否可以拉rtmp流，通过webrtc播放

现在的主播没有美颜根本没法看，而且主播大部分用的都是三星的手机直播，如果加上美颜效果，这个框架肯定会特别受欢迎。

Helo

How i can setting buffer player, i want set buffer to 100, but no properties to set this...

Getting crashes when uninitialize GuestKit.

我在用某app的rtmp做拉流测试的时候 碰到
else if (type == SRS_RTMP_TYPE_SCRIPT) {
if (!srs_rtmp_is_onMetaData(type, data, size)) {
LOG(LS_ERROR) << "No flv";
srs_human_trace("drop message type=%#x, size=%dB", type, size);
}
}
srs_rtmp_is_onMetaData 返回否，请问如何解决 谢谢！

问下是不是有特殊的需求？？
rtmp 地址我用其他的直播 SDK 测试是 OK的

还是这个和其他的 直播SDK, 在对 rtmp URL 上有不同的要求
log如下

2016-10-13 17:07:16.502107 AnyRtmp[9381:1595552] OnRtmpStreamReconnecting:1
2016-10-13 17:07:16.508846 AnyRtmp[9381:1595552] OnRtmpStreamReconnecting:2
2016-10-13 17:07:16.514094 AnyRtmp[9381:1595552] OnRtmpStreamReconnecting:3
2016-10-13 17:07:16.517658 AnyRtmp[9381:1595552] OnRtmpStreamFailed:1

希望能够得到回复, 谢谢

android拉流时视频没有显示出来

Is there any way to mute Rtmplayer volume ?

AnyRTC团队，我在测试Android端推送的时候使用MediaCodec硬件编码，遇到在三星S6、nexus5（都是Android6.0）上一直只输出 I 帧，没有P帧，在三星note3、荣耀1、红米note1（Android5.0及以下版本）上是正常的，将nexus5从Android6.0降到5.0也正常了。
调试时发现只要执行过b.putInt(MediaCodec.PARAMETER_KEY_REQUEST_SYNC_FRAME, 0); 如上两个6.0的机型就再也不输出P帧了，全部都是关键帧。5.0/4.4的机型执行一次输出一个I 帧，不执行就输出P帧，很正常。
本人没找到解决方案，望各位专业大牛帮解决，我的联系QQ：420306380
谢谢。

我的视频数据来自一台专业视频采集设备，可以提供H2.64视频流，请问使用这个库怎么直接推送H2.64视频流数据？

android端有使用教程吗？如何安卓如何用这个来看HLS视屏?

怎么把这个ios推流项目配置到一个新的工程里面?自己配置老是报错.

Build command failed.
Error while executing process D:\PlatformSDKs\android-sdk-windows\ndk-bundle\ndk-build.cmd with arguments {NDK_PROJECT_PATH=null APP_BUILD_SCRIPT=D:\AS_Project\anyRTC-RTMP-OpenSource-master\Prj-Android\jni\toolchain\Android.mk APP_ABI=armeabi NDK_ALL_ABIS=armeabi NDK_DEBUG=1 APP_PLATFORM=android-16 NDK_OUT=D:/AS_Project/anyRTC-RTMP-OpenSource-master/Prj-Android/app/build/intermediates/ndkBuild/debug/obj NDK_LIBS_OUT=D:\AS_Project\anyRTC-RTMP-OpenSource-master\Prj-Android\app\build\intermediates\ndkBuild\debug\lib NDK_APPLICATION_MK:=../jni/Application.mk D:/AS_Project/anyRTC-RTMP-OpenSource-master/Prj-Android/app/build/intermediates/ndkBuild/debug/obj/local/armeabi/libanyrtmp-jni.so}
Android NDK: WARNING:D:/AS_Project/anyRTC-RTMP-OpenSource-master/Prj-Android/jni/toolchain/../Android.mk:anyrtmp-jni: non-system libraries in linker flags: -lavformat -lavcodec -lavutil
Android NDK: This is likely to result in incorrect builds. Try using LOCAL_STATIC_LIBRARIES
Android NDK: or LOCAL_SHARED_LIBRARIES instead to list the library dependencies of the
Android NDK: current module
Android NDK: WARNING: Unsupported source file extensions in D:/AS_Project/anyRTC-RTMP-OpenSource-master/Prj-Android/jni/toolchain/../../../webrtc/Android.mk for module webrtc
Android NDK: system_wrappers/source/trace_impl.h
[armeabi] Compile++ thumb: webrtc <= event_timer_posix.cc
D:/AS_Project/anyRTC-RTMP-OpenSource-master/Prj-Android/jni/toolchain/../../../webrtc/system_wrappers/source/event_timer_posix.cc: In constructor 'webrtc::EventTimerPosix::EventTimerPosix()':
D:/AS_Project/anyRTC-RTMP-OpenSource-master/Prj-Android/jni/toolchain/../../../webrtc/system_wrappers/source/event_timer_posix.cc:53:56: error: 'pthread_condattr_setclock' was not declared in this scope
pthread_condattr_setclock(&cond_attr, CLOCK_MONOTONIC);

我的是一台Android嵌入式设备，使用的是USB摄像头，本想着要传YUV视频数据然后推流，没想到居然能直接支持！！！

anyRTC-RTMP底层使用的是什么技术？为什么可以直接驱动USB摄像头呢？

我用手机的音量键调节不起作用😭

您好，请问我在使用AnyRTC进行推流的过程中，虽然设置了码率1024k和帧率30，但是从实际播放端接收到的效果看，实际码率在200k左右，帧率在8帧左右，请问这是什么原因导致的呢？？

不停的报这个错误！！！
E/SRS: [2017-12-13 02:49:29.40] E/SRS: ignore duplicated sps, code=3044 E/SRS: [2017-12-13 02:49:29.41] E/SRS: ignore duplicated pps, code=3045

I noticed in Android version (6, XIaomi Node), the RTMP stream has a GOP duration of 20 seconds, which cause long latency if enable GOP cache. Otherwise, there's long waiting time for the keyframe when starting play if we turn off GOP cache.

I can't find an API to set KEY_FRAME_INTERVAL parameter anywhere. Please advice.

Thanks

WARNING:C:/Users/etenel/Desktop/anyRTC-RTMP-OpenSource-master/Prj-Android/jni/toolchain/../Android.mk:anyrtmp-jni: non-system libraries in linker flags: -lavformat -lavcodec -lavutil
Android NDK: This is likely to result in incorrect builds. Try using LOCAL_STATIC_LIBRARIES
Android NDK: or LOCAL_SHARED_LIBRARIES instead to list the library dependencies of the
Android NDK: current module
D:/AndroidStudio/SDK/ndk-bundle/build//../toolchains/aarch64-linux-android-4.9/prebuilt/windows-x86_64/bin/../lib/gcc/aarch64-linux-android/4.9.x/../../../../aarch64-linux-android/bin/ld.exe: cannot find -lavformat
D:/AndroidStudio/SDK/ndk-bundle/build//../toolchains/aarch64-linux-android-4.9/prebuilt/windows-x86_64/bin/../lib/gcc/aarch64-linux-android/4.9.x/../../../../aarch64-linux-android/bin/ld.exe: cannot find -lavcodec
D:/AndroidStudio/SDK/ndk-bundle/build//../toolchains/aarch64-linux-android-4.9/prebuilt/windows-x86_64/bin/../lib/gcc/aarch64-linux-android/4.9.x/../../../../aarch64-linux-android/bin/ld.exe: cannot find -lavutil
collect2.exe: error: ld returned 1 exit status
make: *** [C:/Users/etenel/Desktop/anyRTC-RTMP-OpenSource-master/Prj-Android/app/build/intermediates/ndkBuild/debug/obj/local/arm64-v8a/libanyrtmp-jni.so] Error 1
还有日志显示webrtc依赖有问题
Android NDK: Aborting (set APP_ALLOW_MISSING_DEPS=true to allow missing dependencies) . Stop.

请问一下，在用iOS avcaptrue的时候，如果想本地保存每个ts分片怎么办？而且不想通过现保存成mp4再转ts的办法？

想学习Android端直播编码，可以用这个项目来展开么，求教！

hi，
请问支持推流+拉流同时进行吗？因为我想做一个可以互动的app

多谢~~

你好 我如果不想使用gpuimage 需要怎么处理？

我下载的demo中代码如上，现在是 not support。

推流端在分辨率变成一半 比如640480 变成 320240 是什么问题，目前只知道屏幕旋转时可能会出现？请问一下如何解决？

wifi 连接中断 再次连接上如何进行 推流

Thanks for provide this great library.

I'd like to use your library to build video conference app for my local company.

I've tried to build your library and tested. It works perfect but didn't work on arm64, x86 android devices.

I've checked the static library which were used on this library such as ffmpeg, but seems it doesn't build for arm64, x86.

I would appreciate if you let me know how can I use this library for arm64, x86 devices.
Also let me know about license of this library.

Hope to hearing from you soon.
Regards,
gstream

怎么设置w:h=16:9？

请不要打着开源的旗子搞拿来主义，连文档都抄袭。
抄就不说了，抄了还乱改。“集优化H.264软/硬件编码器”这句什么意思？本人原文里写的是“NEON指令集优化H.264软件编码器”，你们这句“集优化”是什么意思？“NEON指令集”是一个整词，不要贻笑大方，这么多人看着呢。

粗看了一下，HEVC/ VP9好像都不支持？

在gpuimage处理完之后的回调函数processVideo中,在调用libyuv::ARGBToI420之前,我把图像数据都设置为0,或者255.
memset([imageFramebuffer byteBuffer],0,width*height*4);
memset([imageFramebuffer byteBuffer],255,width*height*4);

当设为0的情况下,上流出来的颜色是 0x101010
当设为255的情况下,上流出来的颜色是0xeeeeee
这是为什么呢,怎么才能不失真.

For group conference implementation, I've used this library.

When I used only 1 Hosterkit and Guesterkit, it works well. But when I tried to use 2 GuesterKit for pull video from 2 side simulateneously, 1 GuesterKit doesn't work.

Is there any trick to fix this issue ?

Any help would be really appreciated.

Thanks

不支持手机4G网络下推流或拉流吗，目前测试只有wifi可以。

android推流端在旋转屏幕时分辨率发生变化 .MediaCodecVideoEncoder

目前代码VideoCapture.mm:processVideo方法中，直接从[imageFramebuffer byteBuffer]中获取出来使用，这是不是不够完备？没有显示调用glFinish 会导致 cvpixbuffer 获取的是黑屏数据，而如果显示调用glFInish是耗时的。

Helo

I try using nginx-RTMP, Amazing, low latency is OK, but i have problem:

Video packet delay for first time, audio and video not syncronize

TIA

请问Android平台可以一边推流一边录像吗？

Greetings. Thanks for providing this great library.

I am going to use this library for my video conference app.

RTMPHosterKit provide following apis.
-setAudioEnable ()
-setVideoEnable()

setAudioEnable() works great but setVideoEnable() is not working.

Seems it is not implemented yet.

I would appreciate if you let me know how to block video data when streaming or use setVideoEnable() api if it is already implemented.

Thanks
gstream

报这个错
AnyRTC-RTMP\webrtc\base\atomicops.h
Error:(63, 31) error: '__ATOMIC_ACQUIRE' was not declared in this scope
Error:(63, 47) error: '__atomic_load_n' was not declared in this scope
Error:(66, 32) error: '__ATOMIC_RELEASE' was not declared in this scope
Error:(66, 48) error: '__atomic_store_n' was not declared in this scope
Error:(74, 33) error: '__ATOMIC_ACQUIRE' was not declared in this scope

iOS版本，如何实现横屏推流？

相当于一个视频通话的功能，请问这部分集成的话难度有多大？

请问下android端开播现在是否支持美颜滤镜？貌似只看到ios的有集成gpuimage？

我两台机子编译完成下载之后如何去操作推流和拉流

发现iOS运行时 cpu基本占满100% 如何解决？

请问下音频处理（降噪、回音消除等）为何使用android系统自身api而不使用webrtc的apm模块实现呢？貌似android系统自身的api在很多手机上感觉没啥效果。