pcm

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: Someone could tell me why this happen? iOS10, iPhone5s, VOIP Siphon pjsip2.5.5. These issue was not appear every time,it came sometimes when you want to make call or answer call (add at:20161226). Error opening sound device: 14:06:03.137742 pjsua_aud.c ....Set sound device: capture=-1, playback=-2 14:06:03.137775 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms 14:06:03.137837 coreaudio_dev. ......Using VoiceProcessingIO audio unit 14:06:03.147404 acquire_call enter,call_id:2 14:06:03.233365 level:4,msg:14:06:03.233

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I am playing a single channel audio in non-interleaved mode. I am getting underrun when I am writing aduio data into speaker : ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred Here is how I write: printf("%d",snd_pcm_avail (spkhandle)); ret = snd_pcm_writen(spkhandle, pSpeakerBuf , framesIn18Millisec); if(ret < 0) { snd_pcm_recover(spkhandle, ret, 0); } What are the different ways/parameter configurations to prevent ALSA under run ? (I am using Linux 3.0, ARM ) Edit: Here is a buffer measurement using snd_pcm_avail() API snd_pcm_avail

I have an array of audio data I am passing to a reader: recorder.read(audioData,0,bufferSize); The instantiation is as follows: AudioRecord recorder; short[] audioData; int bufferSize; int samplerate = 8000; //get the buffer size to use with this audio record bufferSize = AudioRecord.getMinBufferSize(samplerate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT)*3; //instantiate the AudioRecorder recorder = new AudioRecord(AudioSource.MIC,samplerate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,bufferSize); recording = true; //variable to use

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I am successfully using MediaCodec to decode audio, however when I load a file with 24-bit samples, I have no way of knowing this has occurred. Since the application was assuming 16-bit samples, it fails. When I print the MediaFormat, I see {mime=audio/raw, durationUs=239000000, bits-format=6, channel-count=2, channel-mask=0, sample-rate=96000} I assume that the "bits-format" would be a hint, however this key is not declared in the API, and is not actually emitted when the output format changes. I get {mime=audio/raw, what=1869968451,

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I'm modifying some existing sound code and noticed that when it finishes writing configuration data to the hardware, the code doesn't call snd_pcm_hw_params_free() . The application, for legacy reasons, open and closes the sound hardware for every sound it plays. This has got to be causing memory leak because the snd_pcm_hw_params_t * is never being free'd. So I added a call to `snd_pcm_hw_params_free() and now get the following error: [root@n00200C709F3D namb2]# ./freetest *** glibc detected *** ./freetest: free(): invalid pointer:

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: hey guys i'm trying to run a basic python speech to text code. This is the code. import speech_recognition as sr r = sr.Recognizer() with sr.Microphone() as source: audio = r.listen(source) try: print("You said " + r.recognize(audio)) except LookupError: print("Could not understand audio") The code works fine till it reaches the print stage and then throws this error. Is there anything that i have done wrong? ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2266:(snd_pcm_open_noupdate) Unknown PCM cards