C - как решить эту ошибку при использовании pjsip?

Question

Я получаю эту ошибку при использовании pjsip. Как это решить? У меня есть микрофон/динамик в системе, но он не может получить устройство.

http://paste.ubuntu.com/6504337/

/* Create audio device parameter to open the device */ 
static pj_status_t create_aud_param(pjmedia_aud_param *param, 
        pjmedia_aud_dev_index capture_dev, 
        pjmedia_aud_dev_index playback_dev, 
        unsigned clock_rate, 
        unsigned channel_count, 
        unsigned samples_per_frame, 
        unsigned bits_per_sample) 
{ 
    pj_status_t status; 

    /* Normalize device ID with new convention about default device ID */ 
    if (playback_dev == PJMEDIA_AUD_DEFAULT_CAPTURE_DEV) 
    playback_dev = PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV; 

    /* Create default parameters for the device */ 
    status = pjmedia_aud_dev_default_param(capture_dev, param); 
    if (status != PJ_SUCCESS) { 
    pjsua_perror(THIS_FILE, "Error retrieving default audio " 
       "device parameters", status); 
    return status; 
    } 
    param->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK; 
    param->rec_id = capture_dev; 
    param->play_id = playback_dev; 
    param->clock_rate = clock_rate; 
    param->channel_count = channel_count; 
    param->samples_per_frame = samples_per_frame; 
    param->bits_per_sample = bits_per_sample; 

    /* Update the setting with user preference */ 
#define update_param(cap, field) \ 
    if (pjsua_var.aud_param.flags & cap) { \ 
     param->flags |= cap; \ 
     param->field = pjsua_var.aud_param.field; \ 
    } 
    update_param(PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING, input_vol); 
    update_param(PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, output_vol); 
    update_param(PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE, input_route); 
    update_param(PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE, output_route); 
#undef update_param 

    /* Latency settings */ 
    param->flags |= (PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY | 
      PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY); 
    param->input_latency_ms = pjsua_var.media_cfg.snd_rec_latency; 
    param->output_latency_ms = pjsua_var.media_cfg.snd_play_latency; 

    /* EC settings */ 
    if (pjsua_var.media_cfg.ec_tail_len) { 
    param->flags |= (PJMEDIA_AUD_DEV_CAP_EC | PJMEDIA_AUD_DEV_CAP_EC_TAIL); 
    param->ec_enabled = PJ_TRUE; 
    param->ec_tail_ms = pjsua_var.media_cfg.ec_tail_len; 
    } else { 
    param->flags &= ~(PJMEDIA_AUD_DEV_CAP_EC|PJMEDIA_AUD_DEV_CAP_EC_TAIL); 
    } 

    /* VAD settings */ 
    if (pjsua_var.media_cfg.no_vad) { 
    param->flags &= ~PJMEDIA_AUD_DEV_CAP_VAD; 
    } else { 
    param->flags |= PJMEDIA_AUD_DEV_CAP_VAD; 
    param->vad_enabled = PJ_TRUE; 
    } 

    return PJ_SUCCESS; 
} 

ОШИБКА:

14:13:41.786 pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006] 
Exception: Object: {Account <sip:192.168.1.16:60791>}, operation=make_call(), error=Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) 

EDIT:

Проверьте, что моя система имеет воспроизведения и устройство захвата или нет (как вы можете увидеть следующие шоу 100 % его работы без pjsip):

sun@sun-M14xR2:/var/tmp/pjproject-2.1.0/pjsip/src/pjsua-lib$ aplay -l 
**** List of PLAYBACK Hardware Devices **** 
card 0: PCH [HDA Intel PCH], device 0: CA0132 Analog [CA0132 Analog] 
    Subdevices: 1/1 
    Subdevice #0: subdevice #0 
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] 
    Subdevices: 0/1 
    Subdevice #0: subdevice #0 
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] 
    Subdevices: 1/1 
    Subdevice #0: subdevice #0 

sun@sun-M14xR2:/var/tmp/pjproject-2.1.0/pjsip/src/pjsua-lib$ cat /proc/asound/cards 
0 [PCH   ]: HDA-Intel - HDA Intel PCH 
         HDA Intel PCH at 0xd2710000 irq 47 
1 [NVidia   ]: HDA-Intel - HDA NVidia 
         HDA NVidia at 0xd1000000 irq 17 
2 [U0x46d0x825 ]: USB-Audio - USB Device 0x46d:0x825 
         USB Device 0x46d:0x825 at usb-0000:00:14.0-4, high speed 


sun@sun-M14xR2:/var/tmp/pjproject-2.1.0/pjsip/src/pjsua-lib$ gst-launch-0.10 -v alsasrc device=hw:2 ! audioresample ! audio/x-raw-int,rate=48000 ! autoaudiosink 
Setting pipeline to PAUSED ... 
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000 
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000 
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
Pipeline is live and does not need PREROLL ... 
Setting pipeline to PLAYING ... 
New clock: GstAudioSrcClock 
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink.GstProxyPad:proxypad0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 1.000000 
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE 
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 1.000000 
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE 
WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough 
Additional debug info: 
gstbaseaudiosrc.c(840): gst_base_audio_src_create(): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: 
Dropped 10560 samples. This is most likely because downstream can't keep up and is consuming samples too slowly. 
WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough 
Additional debug info: 
gstbaseaudiosrc.c(840): gst_base_audio_src_create(): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: 
Dropped 9600 samples. This is most likely because downstream can't keep up and is consuming samples too slowly. 

Answer 1

Ваша проблема связана с аудиосистемой. В большинстве систем Linux работает PulseAudio по сравнению с Alsa, как и у вас (вы можете видеть это в журналах GStreamer), но по умолчанию pjsip включает драйвер PortAudio для Linux.

Чтобы исправить это, вы можете включить доступный драйвер Alsa, добавив:

#define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 
#define PJMEDIA_AUDIO_DEV_HAS_ALSA 1 

Для pjlib/включить/р ^/config_site.h. Если он не существует, вы можете создать его любит:

#define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 
#define PJMEDIA_AUDIO_DEV_HAS_ALSA 1 
#include <pj/config_site_sample.h> 

И перестраивать (вы можете непосредственно восстановить только pjmedia: бег make на pjmedia/построить папку).

Примечание: Это может быть возможно, что вы, чтобы отключить текущей конфигурации драйвера путем редактирования pjmedia/build/os-linux.mak и установки AC_PJMEDIA_SND другое значение (например, ALSA)