webrtc_runtime

webrtc_runtime

1. android/ 安卓的编译环境，编译arm时记得在jni/Android.mk去掉下面几行 ../../webrtc/common_audio/fir_filter_sse.cc \

   ../../webrtc/common_audio/resampler/sinc_resampler_sse.cc \

   ../../webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc \

   ../../webrtc/modules/audio_processing/aec/aec_core_sse2.cc \

2. audio/ 测试用的音源

3. build/webrtc_mac/ mac编译环境

4. webrtc/ webrtc 源码，从https://chromium.googlesource.com/external/webrtc/+/branch-heads/61 抽出

5. WebRTCDemo/ unity3d 测试Demo

6. CMakeLists.txt win64 makelist

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WebRTCDemo 运行时要注意的地方

WebRTCDemo 中的AudioDefine.cs里有一系列的设置

其中 DEBUG_NO 默认为 3，这种模式下，只消除网络话音（不能消除背景音乐）

DEBUG_NO = 4 时，要将AudioManager的SampleRate设置为48000，并上手机上测试

MS_IN_SEND_CARDBUF 是消除效果的关键

MS_IN_SEND_CARDBUF = delay

它的算法是 delay = (t_render - t_analyze) + (t_process - t_capture) where:

• t_analyze is the time a frame is passed to AnalyzeReverseStream() and t_render is the time the first sample of the same frame is rendered by the audio hardware.
• t_capture is the time the first sample of a frame is captured by the audio hardware and t_pull is the time the same frame is passed to ProcessStream().

Sets the |delay| in ms between AnalyzeReverseStream() receiving a far-end frame and ProcessStream() receiving a near-end frame containing the corresponding echo. On the client-side this can be expressed as delay = (t_render - t_analyze) + (t_process - t_capture)

where,

t_analyze is the time a frame is passed to AnalyzeReverseStream() and t_render is the time the first sample of the same frame is rendered by the audio hardware. t_capture is the time the first sample of a frame is captured by the audio hardware and t_pull is the time the same frame is passed to ProcessStream().

安卓下 MS_IN_SEND_CARDBUF = 95; 这个值在不同安卓机型上要自己微调

IOS下 MS_IN_SEND_CARDBUF = 65;

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ios10设备，没麦克风权限的解决方案

1.在项目中找到info.plist 文件，右键点击以 Source Code形式打开

2.添加以下键值对 Privacy - Microphone Usage Description 是否允许此App使用你的麦克风？