我想在AppRTCMobile中添加一个接口,这个接口可以启动webrtc Call模块,以实现两部手机之间的音频通话(局域网,已经知道IP地址和端口号),但是当我运行成功时,软件当 RtcEventLog 调用该方法时,每次发生异常时都会崩溃。我不知道Calling Call是否合理。在没有解决方案的情况下,我真诚地感谢您的帮助。下面是源代码,请帮我找出问题所在。
std::unique_ptr<RtcEventLog> event_log = webrtc::RtcEventLog::Create();
webrtc::Call::Config callConfig = webrtc::Call::Config(event_log.get());
callConfig.bitrate_config.max_bitrate_bps = 500*1000;
callConfig.bitrate_config.min_bitrate_bps = 100*1000;
callConfig.bitrate_config.start_bitrate_bps = 250*1000;
webrtc::AudioState::Config audio_state_config = webrtc::AudioState::Config();
cricket::VoEWrapper* g_voe = nullptr;
rtc::scoped_refptr<webrtc::AudioDecoderFactory> g_audioDecoderFactory;
g_audioDecoderFactory = webrtc::CreateBuiltinAudioDecoderFactory();
g_voe = new cricket::VoEWrapper();
audio_state_config.audio_processing = webrtc::AudioProcessing::Create();
g_voe->base()->Init(NULL,audio_state_config.audio_processing,g_audioDecoderFactory);
audio_state_config.voice_engine = g_voe->engine();
audio_state_config.audio_mixer = webrtc::AudioMixerImpl::Create();
callConfig.audio_state = AudioState::Create(audio_state_config);
std::unique_ptr<RtcEventLog> event_logg = webrtc::RtcEventLog::Create();
callConfig.event_log = event_logg.get();
g_call = webrtc::Call::Create(callConfig);
g_audioSendTransport = new AudioLoopbackTransport();
webrtc::AudioSendStream::Config config(g_audioSendTransport);
g_audioSendChannelId = g_voe->base()->CreateChannel();
config.voe_channel_id = g_audioSendChannelId;
g_audioSendStream = g_call->CreateAudioSendStream(config);
webrtc::AudioReceiveStream::Config AudioReceiveConfig;
AudioReceiveConfig.decoder_factory = g_audioDecoderFactory;
g_audioReceiveChannelId = g_voe->base()->CreateChannel();
AudioReceiveConfig.voe_channel_id = g_audioReceiveChannelId;
g_audioReceiveStream = g_call->CreateAudioReceiveStream(AudioReceiveConfig);
g_audioSendStream->Start();
g_audioReceiveStream->Start();