diff --git a/AnyCore/anyrtmpush.cc b/AnyCore/anyrtmpush.cc index d9e32467e7a234e8093e95a195cfc556156180e3..256d22c1057478a89e8bf49a172b33b908a186f6 100644 --- a/AnyCore/anyrtmpush.cc +++ b/AnyCore/anyrtmpush.cc @@ -506,7 +506,9 @@ void AnyRtmpPush::DoSendData() } else if(dataPtr->_type == META_DATA){ int ret = srs_rtmp_write_packet(rtmp_, SRS_RTMP_TYPE_SCRIPT, dataPtr->_dts, (char*)dataPtr->_data, dataPtr->_dataLen); - srs_human_trace("send metadata failed. ret=%d", ret); + if (ret != 0) { + srs_human_trace("send metadata failed. ret=%d", ret); + } return; } diff --git a/Prj-Android/jni/Application.mk b/Prj-Android/jni/Application.mk index 157d6c3707d3041f080f75f11768bdf09d0a8e77..41e50a891466084343a9a1619732c56a14c1eafb 100644 --- a/Prj-Android/jni/Application.mk +++ b/Prj-Android/jni/Application.mk @@ -4,12 +4,12 @@ APP_ABI := armeabi-v7a #armeabi-v7a arm64-v8a NDK_PATH := /cygdrive/c/Android/NDK/android-ndk-r10e -NDK_STL_INC := $(NDK_PATH)/sources/cxx-stl/gnu-libstdc++/4.8/include +NDK_STL_INC := $(NDK_PATH)/sources/cxx-stl/gnu-libstdc++/4.9/include APP_OPTIM := release APP_CFLAGS += -O3 #APP_STL := gnustl_shared APP_STL := gnustl_static -NDK_TOOLCHAIN_VERSION = 4.8 +NDK_TOOLCHAIN_VERSION = 4.9 APP_PLATFORM := android-14 \ No newline at end of file diff --git a/README.md b/README.md index f8d7b70f03830adedacadd4606740f73ab3090ba..4b7804571d715bbd2b97fd58c16edcfe3b5d9992 100644 --- a/README.md +++ b/README.md @@ -1,5 +1,5 @@ # AnyRTC-RTMP - +AnyRTC-RTMP
本次开源的客户端基于RTMP协议的推流拉流客户端,由我司CTO亲自操刀设计,采用跨平台架构一套代码支持Android、iOS、Windows等平台。
直播涉及的流程:『采集->编码->传输->解码->播放』本项目统统包含,这不是软文,这是实实在在的商业级实战代码;无论是你新手还是老司机,我们都热烈欢迎您前来筑码。 diff --git a/webrtc/Android.mk b/webrtc/Android.mk index 4c81a414297d27bb95a9749a3f6e7e390e4b0438..f3f2efc13066579113bb3886c1cc031b613d9076 100644 --- a/webrtc/Android.mk +++ b/webrtc/Android.mk @@ -137,7 +137,7 @@ LOCAL_MODULE := libwebrtc LOCAL_MODULE_TAGS := optional LOCAL_CFLAGS := -std=gnu++11 -frtti -D__UCLIBC__ -DWEBRTC_POSIX -DWEBRTC_LINUX -DWEBRTC_ANDROID -D__STDC_FORMAT_MACROS -D__STDC_CONSTANT_MACROS -D__STDC_LIMIT_MACROS -LOCAL_CFLAGS += -DWEBRTC_THREAD_RR -DWEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE -DWEBRTC_USE_H264 -DWEBRTC_INITIALIZE_FFMPEG +LOCAL_CFLAGS += -DWEBRTC_THREAD_RR -DWEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE -DWEBRTC_USE_H264 -DWEBRTC_INITIALIZE_FFMPEG -DNO_STL include $(BUILD_STATIC_LIBRARY)