Snd_pcm_open Error Codes
The underrun can happen when an application does not feed new samples in time to alsa-lib (due CPU usage). General overview ALSA uses the ring buffer to store outgoing (playback) and incoming (capture, record) samples. ALSA Duplex Working code Submitted by Santoshkumar (not verified) on Fri, 09/26/2008 - 07:24. Example: hw hw:0 hw:0,0 hw:supersonic,1 hw:soundwave,1,2 hw:DEV=1,CARD=soundwave,SUBDEV=2 Plug->HW device The plughw device description uses the plug plugin and hw plugin as slave. click site
Transfer methods in UNIX environments In the UNIX environment, data chunk acknowledges are received via standard I/O calls or event waiting routines (poll or select function). Parameters: pcmPCM handle bufsframes containing buffers (one for each channel) sizeframes to be written Returns:a positive number of frames actually written otherwise a negative error code Return values: -EBADFDPCM is not int snd_pcm_hw_free ( snd_pcm_t * pcm) Remove PCM hardware configuration and free associated resources. ch 2(2) sr 48000 (reenc 0) 2010-10-28 21:53:20.728 Opening ALSA audio device 'default'.
Examples: /test/pcm.c. The opposite function is snd_pcm_unlink(). For processing all pending samples, use ::snd_pcm_drain() instead. I will stay with OSS until ALSA becomes much more user-friendly.
- It is brilliant tutorial for beginners.I have run the capture and playback code.I noticed my volume level is very low when I capture my voice followed by playback code.But if I
- Problem in opening default device in listing2 of this article Submitted by Nagaraja S (not verified) on Fri, 03/11/2005 - 05:56.
- These parameters can be obtained: the current stream state - snd_pcm_status_get_state(), timestamp of trigger - snd_pcm_status_get_trigger_tstamp(), timestamp of last pointer update snd_pcm_status_get_tstamp(), delay in samples - snd_pcm_status_get_delay(), available count in samples
- It also checks if the protocol is compatible to prevent the use of programs written to an older API with newer drivers.
- You'll need this handle when you call the other snd_pcm_* functions.
- Typically, you would add the option -lasound on the linker command line.
- It means, for playback, the empty samples in ring buffer and for capture, the filled (used) samples in ring buffer.
- The usual place for default definitions is at /usr/share/alsa/alsa.conf.
- It is as such the overall latency from the initial ADC to the read call.
- If you had any user data associated with it, you can get to it like this: void *private_data = snd_async_handler_get_callback_private(pcm_callback); Where 'void *' would, of course, be whatever type the original
Some ALSA library functions use the dlopen function and floating-point operations, so you also may need to add -ldl and -lm. The function is thread-safe when built with the proper option. Web browsing is one of the common processes that attracts the occurrence of these errors. Alsa Error Codes User contributions on this site are licensed under the Creative Commons Attribution Share Alike 4.0 International License.
great article tho :) i am working on a project Submitted by Anonymous (not verified) on Thu, 10/09/2008 - 22:38. Snd_pcm_nonblock Firstly, this article is really good. After having tested it on a number of boards, it was always a pain in the neck to configure properly, and getting it to work, an adventure. http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html Sequencer interface: a higher-level interface for MIDI programming and sound synthesis than the raw MIDI interface.
intsnd_pcm_wait (snd_pcm_t *pcm, int timeout) Wait for a PCM to become ready. Alsa Playback Example the mode in which to open the device. Default device The default device is equal to plug plugin with hw plugin as slave. 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
for code called 'alsa.c' it's gcc alsa.c -lasound Linking Libraries Submitted by Elie (not verified) on Mon, 12/29/2008 - 23:53. http://tomdownload.net/software/snd_pcm_open-error-codes/ If everything goes as planned, the application should display the following messages, or something similar:Audio device opened successfully. Snd_pcm_open Example Errors: -ENOMEM Not enough memory to allocate control structures. Snd_pcm_hw_params_set_channels This is where our callback comes in.
intsnd_pcm_open_fallback (snd_pcm_t **pcm, snd_config_t *root, const char *name, const char *orig_name, snd_pcm_stream_t stream, int mode) Opens a fallback PCM. get redirected here Returns: Zero on success, or a negative error code. When we open the PCM stream, we specify the mode as SND_PCM_STREAM_CAPTURE. It gives /opt/arm-2009q1/bin/../lib/gcc/arm-none-linux-gnueabi/4.3.3/../../../../arm-none-linux-gnueabi/bin/ld: cannot find -lasound collect2: ld returned 1 exit status what is my mistake? Snd_pcm_hw_params_set_rate_near
In this case, we tell it to wait until our buffer is almost full. For stopping the PCM stream immediately, use ::snd_pcm_drop() instead. Parameters: pcmPCM handle availpNumber of available frames in the ring buffer delaypTotal I/O latency in frames Returns:zero on success otherwise a negative error code The avail and delay values retuned are http://dualathlonserver.com/error-codes/sick-pls-error-codes.php int snd_pcm_poll_descriptors_count ( snd_pcm_t * pcm) get count of poll descriptors for PCM handle Parameters: pcmPCM handle Returns:count of poll descriptors The function is thread-safe when built with the proper option.
Parameters: pcmPCM handle availNumber of available frames when timestamp was grabbed tstampHi-res timestamp Returns:0 on success otherwise a negative error code Note this function does not update the actual r/w pointer Snd_pcm_set_params It worked a treat. device The audio device number.
The function is thread-safe when built with the proper option.
Nothing is better than knowing how to troubleshoot it by yourself. The function is thread-safe when built with the proper option. This article is simply great, starts from scratch building foundation and then takes you as far as you want to go.. Snd_pcm_t The call to snd_pcm_open opens the default PCM device and sets the access mode to PLAYBACK.
A pointer is maintained to keep track of the current positions in both the hardware buffer and the application buffer. History of ALSA The ALSA Project was started because the sound drivers in the Linux kernel (OSS/Free drivers) were not being maintained actively and were lagging behind the capabilities of new Scott Adv Reply October 29th, 2010 #7 bance View Profile View Forum Posts Private Message Gee! http://dualathlonserver.com/error-codes/smpp-dlr-error-codes.php The value returned by that call is not directly related to the delay, since the latter might include some additional, fixed latencies the former does not.
A pretty common sample format, and it should be obvious how these parameters are named. In Ubuntu, which was the Linux distribution that was used for this tutorial, g++, libasound2, and libasound2-dev are not installed by default and will need to be installed using apt-get (sudo Thanks Manu 5 sec Submitted by MK (not verified) on Thu, 04/09/2009 - 11:06. /* 5 seconds in microseconds divided by * period time */ loops = 5000000 / val; How Pablo Re: Introduction to Sound Programming with ALSA Submitted by Anonymous on Fri, 10/01/2004 - 02:00.
Non-interleaved means data is transfered in periods, where each period is composed of a chunk of samples from each channel. If application wants to manage the ahead samples itself, the snd_pcm_rewind() function allows to forget the last samples in the stream. 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 other major factor affecting sound quality is the sampling rate.
snd_pcm_avail_update()) are thread-safe and can be called concurrently from multiple threads. Adv Reply October 29th, 2010 #2 klc5555 View Profile View Forum Posts Private Message Iced Almond Soy Ubuntu, No Foam Join Date Mar 2008 Beans 1,086 Re: Audio no longer Typedef Documentation typedef struct _snd_pcm_access_mask snd_pcm_access_mask_t PCM access types mask typedef struct _snd_pcm_format_mask snd_pcm_format_mask_t PCM formats mask typedef struct _snd_pcm_hw_params snd_pcm_hw_params_t PCM hardware configuration space container snd_pcm_hw_params_t is an opaque structure Interleaved organization means, that samples from channels are mixed together.
The function 'set_avail_min' tells ALSA when to notify us.