Change some parameters, such as the sampling rate and data format, and see how it affects the results. Two times the period size seems to be a good initial write size? The other major factor affecting sound quality is the sampling rate. Writing a sequencer client The example seqdemo.c demonstrates how to create a simple client for the ALSA sequencer.
Parameters: pcmPCM handle Returns:0 on success otherwise a negative error code The function is thread-safe when built with the proper option. I am using ALSA. The first three can be used for direct communication. This takes exclusive control over the sound devices, so _all_ other programs using ALSA will not work.
The poll desctiptor array should have the size returned by ::snd_pcm_poll_descriptors_count() function. You can check it via ::snd_pcm_hw_params_can_pause() function. The function is thread-safe when built with the proper option. const char* snd_pcm_name ( snd_pcm_t * pcm) get identifier of PCM handle Parameters: pcmPCM handle Returns:ascii identifier of PCM handle Returns the ASCII identifier of given PCM handle.
Examples: /test/pcm.c. Examples: if (ev->data.control.channel < split_channel) channel splitting if (ev->data.note.velocity < threshold) velocity dependence if (ev->type == SND_SEQ_EVENT_CONTROLLER) extract controller events c) int main(int argc, char *argv) Well, this is not essentially The following picture shows a perfect sinus waveform: Next image shows digitized representation: As you may see, the quality of digital audio signal depends on the time (recording rate) and voltage Snd_pcm_writei Example snd_pcm_sframes_t snd_pcm_forward ( snd_pcm_t * pcm, snd_pcm_uframes_t frames ) Move application frame position forward.
cat file-of-random-data > /dev/snd/pcmC0D0p then I recieve what seems to be an error from cat cat: write error: File descriptor in bad state How can I fix this so I can You have a problem with _recording_, right? Change the hardware parameter values and observe how the displayed results change. share|improve this answer answered May 25 '11 at 11:35 jmtd 3,9331621 1 Thanks for the helpful suggestion.
First, stepping through the buffer is the right thing to do; you don't need to write a sample more than once. Snd_pcm_drain Since we would like the sequence to be looped, we schedule a SND_SEQ_EVENT_ECHO event after each sequence. After testing, I had to conclude that the sample code didn't supply enough samples to overcome the device latency before the snd_pcm_close function was called which implies that the close function snd_pcm_sframes_t snd_pcm_rewind ( snd_pcm_t * pcm, snd_pcm_uframes_t frames ) Move application frame position backward.
The field values in pollfd structs may be bogus regarding the stream direction from the application perspective (POLLIN might not imply read direction and POLLOUT might not imply write), but the http://comphelp.org/guide/snd-pcm-write-error-codes/ We set the stream to interleaved mode, 16-bit sample size, 2 channels and a 44,100 bps sampling rate. Alsa Pcm Example The value "default" we use here opens up either the first available device or whatever the default is, probably set in the system or user's asoundrc file. Alsa Error Codes It does not accept any I/O calls in this state.
great article tho :) i am working on a project Submitted by Anonymous (not verified) on Thu, 10/09/2008 - 21:38. For plaback compile code for playback in the seprate folder and run ./.outfile name Converting to ALSA Submitted by Anonymous (not verified) on Sat, 01/31/2009 - 02:19. Notes on writing a GUI based audio application You probably would like to use PCM audio in a GUI based application. When the snd_pcm_link() function is called, all operations managing the stream state for these two streams are joined. Snd_pcm_hw_params_set_channels
If there's not much else to do, there's always the sleep system call. The arguments (in order: FILE,FORMAT) specify filename and file format. How many times do you need to beat mom and Satan etc to 100% the game? how to read & play a sound file Submitted by Anonymous (not verified) on Thu, 07/15/2010 - 03:11.
In the loop that manages data, we read from standard input and fill our buffer with one period of samples. Snd_pcm_wait snd_pcm_drainThe snd_pcm_drain() function enters the SND_PCM_STATE_DRAINING, if the capture device has some samples in the ring buffer otherwise SND_PCM_STATE_SETUP state is entered. snd_pcm_sframes_tsnd_pcm_readn (snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size) Read non interleaved frames to a PCM.
The function snd_pcm_avail() reads the current hardware pointer in the ring buffer from hardware and calls snd_pcm_avail_update() then. http://www.alsa-project.org/alsa-doc/alsa-lib/ The ALSA library API reference. You normally can't write audio using /dev/dsp anymore, at least without being tricky. Snd_pcm_nonblock If you don't like this, you can modify it and let e.g.
Please try the request again. In this case, you have to add the lines if ((ev->type == SND_SEQ_EVENT_NOTEON) && (ev->data.note.velocity == 0)) ev->type = SND_SEQ_EVENT_NOTEOFF; directly after snd_seq_event_input. int snd_pcm_drain ( snd_pcm_t * pcm) Stop a PCM preserving pending frames. For interleaved write access, we use the function /* Write num_frames frames from buffer data to */ /* the PCM device pointed to by pcm_handle. */ /* Returns the number of
wrt testing, FIX your mixer settings. But there are a few problems. This also applies to the access type and the number of channels. We use the period size chosen by ALSA and make this the size of our buffer for storing samples.
Parameters: pcmPCM handle Returns:0 on success otherwise a negative error code This function stops the PCM immediately. Data Structures struct snd_pcm_audio_tstamp_config_t struct snd_pcm_audio_tstamp_report_t struct snd_pcm_channel_area_t union snd_pcm_sync_id_t struct snd_pcm_chmap_t struct snd_pcm_chmap_query_t Macros #defineSND_PCM_DLSYM_VERSION_dlsym_pcm_001 #defineSND_PCM_NONBLOCK #defineSND_PCM_ASYNC #defineSND_PCM_ABORT0x00008000 #defineSND_PCM_NO_AUTO_RESAMPLE0x00010000 #defineSND_PCM_NO_AUTO_CHANNELS0x00020000 #defineSND_PCM_NO_AUTO_FORMAT0x00040000 #defineSND_PCM_NO_SOFTVOL0x00080000 #defineSND_CHMAP_API_VERSION((1 << 16) | (0 << 8) | The second parameter is our device handle again, the third our callback function and the fourth is a void pointer to any user data you may want to pass to the Managing the stream state The following functions directly and indirectly affect the stream state: snd_pcm_hw_paramsThe snd_pcm_hw_params() function brings the stream state to SND_PCM_STATE_SETUP if successfully finishes, otherwise the state SND_PCM_STATE_OPEN is
There's at least one buffer full of data that's being transferred to the card at any given moment, and there's buffering on the application side, which is what you seem to Hopefully, some Developer With A Clue will extend or fix this document. Access SND_PCM_ACCESS_MMAP_COMPLEX does not fit to interleaved and non-interleaved ring buffer organization. This is easy, really: snd_pcm_hw_params (pcm_handle, hw_params); Of course, we also have to clean up the structure behind our back.
The overrun can happen when an application does not take new captured samples in time from alsa-lib. -ESTRPIPE This error means that system has suspended drivers. It would be nice if the (fixed) page width were a little greater, without having to shrink the browser text size. wrt IEC 958 and alsa, possibly an explanation here may help: http://www.alsa-project.org/~tiwai/w...r/ch10s03.html . midiroute.c provides a simple example.