At Mon, 23 Apr 2007 16:18:30 +0200, Nyx wrote: > > Hi, > > I have some problems to read the input of my soundcard using Alsa and I > don't really understand how to have access to the input of my soundcard. > I follow : > http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html > http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.html > > > # So, in my program, I have created two handles for capture and playback : > char *device = "hw:0,0"; > snd_output_t *output; > snd_input_t *input; > snd_pcm_t *phandle, *chandle; > > # then I connect the phandle to the out > err = snd_output_stdio_attach(&output, stdout, 0); > if (..) > > # Do I have to connect the chandle with the input ? > # err = snd_input_stdio_attach(&input, stdin, 0); > > # I open the access to PCM > err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, 0) > if (..) > err = snd_pcm_open(&chandle, pdevice, SND_PCM_STREAM_CAPTURE, 0) > if(..) > > # Hardware and software Parameters > ## Hardware Parameters > snd_pcm_hw_params; > int err; > > err = snd_pcm_hw_params_any(phandle, params); if(..); > err = snd_pcm_hw_params_set_access(phandle, > params,SND_PCM_ACCESS_RW_INTERLEAVED); if(..); > err = snd_pcm_hw_params_set_format(phandle, params,SND_PCM_FORMAT_S16); > if(..); > err = snd_pcm_hw_params_set_channels(phandle, params, 1); if(..); > err = snd_pcm_hw_params_set_rate_near(phandle, params, 44100, 0); if(..); > err = snd_pcm_hw_params_set_buffer_time_near(phandle, params, 500000, &dir); > if(..); > err = snd_pcm_hw_params_set_time_near(phandle, params, 100000, &dir), > if(..); > err = snd_pcm_hw_params(phandle, params); > > ##Software parameters > err = snd_pcm_sw_params(phandle, params); if(..); > err = snd_pcm_sw_params_set_start_threshold(phandle, swparams, 0x7fffffff); > if(..); > err = snd_pcm_sw_params_set_avail_min(phandle, swparams, 4); > > # same configuration for phandle > > # Link output to input and start > err = snd_pcm_link(chandle, phandle); if(..); > err = snd_pcm_start(chandle); if(..); > > #Then I want to read my input > frames_in = 0; > in_max = 0; > latency = 28; > buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2); > while (1) { > r = readbuf(chandle, buffer, latency, &frames_in, &in_max)); if(..); > > } > > # description of the function > long readbuf(snd_pcm_t *handle, char *buf, long len, size_t *frames, size_t > *max) > { > long r; > do { > r = snd_pcm_readi(handle, buf, len); > } while (r == -EAGAIN); > > if (r > 0) { > *frames += r; > if ((long)*max < r) > *max = r; > } > printf ("r = %s, len %li and buf[%d] %f \r", snd_strerror(r), len,i, > buf[i]); > return r; > } > > So I compile it and I run it, and I have > $ read = Broken Pipe, len = 28 and buf[..] = 0.000 > > Anyone can help me to read the input of my sound card ? When you link both playback and capture streams and start them at the same time, then it results in buffer underrun for the playback -- unless you fill the data beforehand. snd_pcm_start() triggers _both_ streams linked together. Hence, a common technique for such full-duplex streams is to fill empty data to the playback stream beforehand, then starts. Takashi _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
