[alsa-devel] Why analog loopback of sigmatel devices got removed?
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Searching around I found this:
http://article.gmane.org/gmane.linux.alsa.devel/59878
* Removed analog loopback mixer items, not mentioned in sigmatel datasheets and reported to break sound support.
First they are mention. It took me lot of time to find them. I only have added the loopback items for devices that have this feature in datasheet.
Then how it can break sound support? could you explain? I need this feature since I use it with my TV card, and I know more users that use that feature.
I don't know about other devices, but I attach the my patch that was included in kernel for this feature.
Best regards, Maxim Levitsky
At Fri, 19 Jun 2009 02:50:35 +0300, Maxim Levitsky wrote:
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Because it harms than helps more often. I've got tons of bug reports and been upset just because users blindly set "Analog Loopback" mixer switch and lost their sound output. This is no mixer element that behaves intuitively -- it shuts out PCM output instead of adding the analog loopback like other hardwares.
The feature is still available, though. You just need to give the driver an instruction via sysfs. Namely, write to sysfs files corresponding to the IDT codec like # echo "loopback yes" > /sys/class/sound/hwC0D0/hints # echo 1 > /sys/class/sound/hwC0D0/reconfig Then the driver will reconfigure with the loopback mixer element. I have a patch series to add this with a "patch" firmware file to make it easier, which I'm going to send an RFC to this ML soon later. Take a look at topic/hda-patch branch on sound-unstable tree.
thanks,
Takashi
Mates...
I am getting a segfault when running my app:
This is the part of the code where I am getting the error:
if ((err = set_swparams(handle, swparams)) < 0) { printf("Setting of swparams failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); }
That is the gdb output: (*The full program code's been posted below the debug output)
Starting program: /home/guilherme/Experimentos/PCM [Thread debugging using libthread_db enabled] [New Thread 0xb7c896c0 (LWP 3391)] [Switching to Thread 0xb7c896c0 (LWP 3391)]
Breakpoint 1, main (argc=1, argv=0xbfa27da4) at pcm.c:777 777 }; (gdb) n 783 int method = 0; (gdb) n 791 morehelp = 0; (gdb) n 794 if ((c = getopt_long(argc, argv, "hd:r:c:f:b:p:m:o:vne", long_option, NULL)) < 0) (gdb) nn Undefined command: "nn". Try "help". (gdb) n 857 if(morehelp) { (gdb) n 863 err = snd_output_stdio_attach(&output, &stdout, 0); (gdb) n 864 if (err < 0) { (gdb) n 869 printf("Playback device is %s\n", device); (gdb) n Playback device is default 870 printf("Stream parameters are %iHz, %s, %i channels\n", rate, snd_pcm_format_name(format), channels); (gdb) n Stream parameters are 48000Hz, S16_LE, 1 channels 871 printf("sine wave rate is %i %.4fHz\n", freq); (gdb) n sine wave rate is 0 0.0000Hz 872 printf("Using transfer method: %s\n", transfer_methods[method].name); (gdb) n Using transfer method: write 874 if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { (gdb) n 879 if ((err = set_hwparams(handle, hwparams, transfer_methods[method].access)) < 0) { (gdb) n
Program received signal SIGSEGV, Segmentation fault. 0xb7d47e3f in memset () from /lib/libc.so.6 (gdb) Quit (gdb) quit The program is running. Exit anyway? (y or n)
That is the code:
#include <stdio.h> #include <stdlib.h> #include <string.h> #include <sched.h> #include <errno.h> #include <getopt.h> #include <alsa/asoundlib.h> #include <sys/time.h> #include <math.h>
static char *device = "default"; //playback device aplay -l static snd_pcm_format_t format = SND_PCM_FORMAT_S16; //sample format unsigned 16 bit endian static unsigned int rate = 48000; //stream rate static unsigned int channels = 1; //contagem dos canais static unsigned int buffer_time = 500; //ring buffer length on us static unsigned int period_time = 100; //period time in us static double freq = 440; // frequencia da senóide em Hz static int verbose = 0; //verbose flag static int resample = 1; //abilitar alsa-lib resampling static int period_event = 0; // produz um event pool a cada periodo
static snd_pcm_sframes_t buffer_size; //quantidade de buffers frames static snd_pcm_sframes_t period_size; //quantidade de periods frames static snd_output_t *output = NULL; //ALSA usa esse ponteiro para lidar com output objects
static void generate_sine(const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, int count, double *_phase) { static double max_phase = 2. * M_PI; double phase = *_phase; double step = max_phase*freq / (double)rate; double res; unsigned char *samples[channels], *tmp; int steps[channels]; unsigned int chn, byte;
union { int i; unsigned char c[4]; } ires; unsigned int maxval = (1 << (snd_pcm_format_width(format) - 1)) - 1; int bps = snd_pcm_format_width(format) / 8; //bits per sample
/*verificar e preparar o conteúdo das areas*/ for (chn = 0; chn < channels; chn++) { if ((areas[chn].first % 8) != 0) { printf("areas[%i].first == %i, abortando...\n", chn, areas[chn].first); exit(EXIT_FAILURE); } //pega endereço e offset da area samples[chn] = /*(signed short *)*/(((unsigned char *)areas[chn].addr) + (areas[chn].first / 8));
if ((areas[chn].step % 16) != 0) { printf("areas[%i].step == %i, aborting...\n", chn, areas[chn].step); exit(EXIT_FAILURE); }
steps[chn] = areas[chn].step / 8; samples[chn] += offset * steps[chn]; }
//completa os channels da area while(count-- > 0) { res = sin(phase) * maxval; ires.i = res; tmp = ires.c; for (chn = 0; chn < channels; chn++) { for (byte = 0; byte < (unsigned int)bps; byte++) *(samples[chn] + byte) = tmp[byte]; samples[chn] += steps[chn]; } phase += step; if (phase >= max_phase) phase -= max_phase; } *_phase = phase;
}
static int set_hwparams(snd_pcm_t *handle, snd_pcm_hw_params_t *params, snd_pcm_access_t access) { unsigned int rrate; snd_pcm_uframes_t size; int err, dir;
/*escolha todos os parametros*/ err = snd_pcm_hw_params_any(handle, params); if (err < 0) { printf("Broken configuration for playback: no configuration available: %s\n", snd_strerror(err)); return err; }
/*set hardware resampling*/ err = snd_pcm_hw_params_set_rate_resample(handle, params, resample); if (err < 0) { printf("Resampling setup failed for playback: %s\n", snd_strerror(err)); return err; }
/* set the interleaved read/write format */ err = snd_pcm_hw_params_set_access(handle, params, access); if (err < 0) { printf("Access type not available for playback: %s\n", snd_strerror(err)); return err; }
/* set the sample format SND_PCM_FORMAT_S16 */ err = snd_pcm_hw_params_set_format(handle, params, format); if (err < 0) { printf("Sample format not available for playback: %s\n", snd_strerror(err)); return err; }
/* set the count of channels 1 */ err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { printf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err)); return err; }
/* set the stream rate */ rrate = rate; err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0); if (err < 0) { printf("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err)); return err; } if (rrate != rate) { printf("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err); return -EINVAL; } /* set the buffer time */ err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir); if (err < 0) { printf("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_buffer_size(params, &size); if (err < 0) { printf("Unable to get buffer size for playback: %s\n", snd_strerror(err)); return err; } buffer_size = size; /* set the period time */ err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_period_size(params, &size, &dir); if (err < 0) { printf("Unable to get period size for playback: %s\n", snd_strerror(err)); return err; } period_size = size; /* write the parameters to device */ err = snd_pcm_hw_params(handle, params); if (err < 0) { printf("Unable to set hw params for playback: %s\n", snd_strerror(err)); return err; } return 0; }
static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams) {
int err;
/*pegar os parametros atuais swparams*/ err = snd_pcm_sw_params_current(handle, swparams); if (err < 0) { printf("Unable to determine current swparams for playback: %s\n", snd_strerror(err)); return err; } /*Começa a transferência quando o buffer está quase cheio*/ /*(buffer_size / avail_min) * avail_min */ err = snd_pcm_sw_params_set_avail_min(handle, swparams, (buffer_size / period_size) * period_size); if (err < 0) { printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err)); return err; } /* allow the transfer when at least period_size samples can be processed */ /* or disable this mechanism when period event is enabled (aka interrupt like style processing) */ err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_event ? buffer_size : period_size); if (err < 0) { printf("Unable to set avail min for playback: %s\n", snd_strerror(err)); return err; } /* enable period events when requested */ if (period_event) { err = snd_pcm_sw_params_set_period_event(handle, swparams, 1); if (err < 0) { printf("Unable to set period event: %s\n", snd_strerror(err)); return err; } } /* write the parameters to the playback device */ err = snd_pcm_sw_params(handle, swparams); if (err < 0) { printf("Unable to set sw params for playback: %s\n", snd_strerror(err)); return err; } return 0; }
/* Tenta recuperar estados de underrun e suspend */
static int xrun_recovery(snd_pcm_t *handle, int err) { if (verbose) printf("stream recovery\n"); if (err == -EPIPE) { //under-run*/ err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) sleep(1); /*espera pela release flag*/ if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err; }
/* Metodo de transferência *WRITE-ONLY* */ static int write_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { double phase = 0; signed short *ptr; int err, cptr;
while(1) { generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_writei(handle, ptr, cptr); if (err == -EAGAIN) continue; if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); }//if ptr += err * channels; cptr -= err; }//if }//if
}//while }
/* * Método de transferência - write and wait por espaço no buffer */
static int wait_for_poll(snd_pcm_t *handle, struct pollfd *ufds, unsigned int count) { unsigned short revents;
while (1) { poll(ufds, count, -1); snd_pcm_poll_descriptors_revents(handle, ufds, count, &revents); if (revents & POLLERR) return -EIO; if (revents & POLLOUT) return 0; } }
static int write_and_poll_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { struct pollfd *ufds; double phase = 0; signed short *ptr; int err, count, cptr, init;
count = snd_pcm_poll_descriptors_count (handle); if (count <= 0) { printf("Invalid poll descriptors count\n"); return count; }
ufds = malloc(sizeof(struct pollfd) * count); if (ufds == NULL) { printf("No enough memory\n"); return -ENOMEM; } if ((err = snd_pcm_poll_descriptors(handle, ufds, count)) < 0) { printf("Unable to obtain poll descriptors for playback: %s\n", snd_strerror(err)); return err; }
init = 1; while (1) { if (!init) { err = wait_for_poll(handle, ufds, count); if (err < 0) { if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN || snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) { err = snd_pcm_state(handle) == SND_PCM_STATE_XRUN ? -EPIPE : -ESTRPIPE; if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; } else { printf("Wait for poll failed\n"); return err; } } }
generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_writei(handle, ptr, cptr); if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; break; /* skip one period */ } if (snd_pcm_state(handle) == SND_PCM_STATE_RUNNING) init = 0; ptr += err * channels; cptr -= err; if (cptr == 0) break; /* it is possible, that the initial buffer cannot store */ /* all data from the last period, so wait awhile */ err = wait_for_poll(handle, ufds, count); if (err < 0) { if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN || snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) { err = snd_pcm_state(handle) == SND_PCM_STATE_XRUN ? -EPIPE : -ESTRPIPE; if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; } else { printf("Wait for poll failed\n"); return err; } } } } }
/* * Metodo de transferencia - Notificação assincrona */
struct async_private_data { signed short *samples; snd_pcm_channel_area_t *areas; double phase; };
static void async_callback(snd_async_handler_t *ahandler) { snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler); struct async_private_data *data = snd_async_handler_get_callback_private(ahandler); signed short *samples = data->samples; snd_pcm_channel_area_t *areas = data->areas; snd_pcm_sframes_t avail; int err;
avail = snd_pcm_avail_update(handle); while (avail >= period_size) { generate_sine(areas, 0, period_size, &data->phase); err = snd_pcm_writei(handle, samples, period_size); if (err < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (err != period_size) { printf("Write error: written %i expected %li\n", err, period_size); exit(EXIT_FAILURE); } avail = snd_pcm_avail_update(handle); } }
static int async_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { struct async_private_data data; snd_async_handler_t *ahandler; int err, count;
data.samples = samples; data.areas = areas; data.phase = 0; err = snd_async_add_pcm_handler(&ahandler, handle, async_callback, &data); if (err < 0) { printf("Unable to register async handler\n"); exit(EXIT_FAILURE); } for (count = 0; count < 2; count++) { generate_sine(areas, 0, period_size, &data.phase); err = snd_pcm_writei(handle, samples, period_size); if (err < 0) { printf("Initial write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (err != period_size) { printf("Initial write error: written %i expected %li\n", err, period_size); exit(EXIT_FAILURE); } } if (snd_pcm_state(handle) == SND_PCM_STATE_PREPARED) { err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } }
/* because all other work is done in the signal handler, suspend the process */ while (1) { sleep(1); } }
/* * Transfer method - asynchronous notification + direct write */
static void async_direct_callback(snd_async_handler_t *ahandler) { snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler); struct async_private_data *data = snd_async_handler_get_callback_private(ahandler); const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t avail, commitres; snd_pcm_state_t state; int first = 0, err;
while (1) { state = snd_pcm_state(handle); if (state == SND_PCM_STATE_XRUN) { err = xrun_recovery(handle, -EPIPE); if (err < 0) { printf("XRUN recovery failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } else if (state == SND_PCM_STATE_SUSPENDED) { err = xrun_recovery(handle, -ESTRPIPE); if (err < 0) { printf("SUSPEND recovery failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } avail = snd_pcm_avail_update(handle); if (avail < 0) { err = xrun_recovery(handle, avail); if (err < 0) { printf("avail update failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; continue; } if (avail < period_size) { if (first) { first = 0; err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } else { break; } continue; } size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } generate_sine(my_areas, offset, frames, &data->phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } size -= frames; } } }
static int async_direct_loop(snd_pcm_t *handle, signed short *samples ATTRIBUTE_UNUSED, snd_pcm_channel_area_t *areas ATTRIBUTE_UNUSED) { struct async_private_data data; snd_async_handler_t *ahandler; const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t commitres; int err, count;
data.samples = NULL; /* we do not require the global sample area for direct write */ data.areas = NULL; /* we do not require the global areas for direct write */ data.phase = 0; err = snd_async_add_pcm_handler(&ahandler, handle, async_direct_callback, &data); if (err < 0) { printf("Unable to register async handler\n"); exit(EXIT_FAILURE); } for (count = 0; count < 2; count++) { size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } generate_sine(my_areas, offset, frames, &data.phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } size -= frames; } } err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); }
/* because all other work is done in the signal handler, suspend the process */ while (1) { sleep(1); } }
/* * Transfer method - direct write only */
static int direct_loop(snd_pcm_t *handle, signed short *samples ATTRIBUTE_UNUSED, snd_pcm_channel_area_t *areas ATTRIBUTE_UNUSED) { double phase = 0; const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t avail, commitres; snd_pcm_state_t state; int err, first = 1;
while (1) { state = snd_pcm_state(handle); if (state == SND_PCM_STATE_XRUN) { err = xrun_recovery(handle, -EPIPE); if (err < 0) { printf("XRUN recovery failed: %s\n", snd_strerror(err)); return err; } first = 1; } else if (state == SND_PCM_STATE_SUSPENDED) { err = xrun_recovery(handle, -ESTRPIPE); if (err < 0) { printf("SUSPEND recovery failed: %s\n", snd_strerror(err)); return err; } } avail = snd_pcm_avail_update(handle); if (avail < 0) { err = xrun_recovery(handle, avail); if (err < 0) { printf("avail update failed: %s\n", snd_strerror(err)); return err; } first = 1; continue; } if (avail < period_size) { if (first) { first = 0; err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } else { err = snd_pcm_wait(handle, -1); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("snd_pcm_wait error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } } continue; } size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } generate_sine(my_areas, offset, frames, &phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } size -= frames; } } }
/* * Transfer method - direct write only using mmap_write functions */
static int direct_write_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { double phase = 0; signed short *ptr; int err, cptr;
while (1) { generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_mmap_writei(handle, ptr, cptr); if (err == -EAGAIN) continue; if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } break; /* skip one period */ } ptr += err * channels; cptr -= err; } } }
struct transfer_method { const char *name; snd_pcm_access_t access; int (*transfer_loop)(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas); };
static struct transfer_method transfer_methods[] = { { "write", SND_PCM_ACCESS_RW_INTERLEAVED, write_loop }, { "write_and_poll", SND_PCM_ACCESS_RW_INTERLEAVED, write_and_poll_loop }, { "async", SND_PCM_ACCESS_RW_INTERLEAVED, async_loop }, { "async_direct", SND_PCM_ACCESS_MMAP_INTERLEAVED, async_direct_loop }, { "direct_interleaved", SND_PCM_ACCESS_MMAP_INTERLEAVED, direct_loop }, { "direct_noninterleaved", SND_PCM_ACCESS_MMAP_NONINTERLEAVED, direct_loop }, { "direct_write", SND_PCM_ACCESS_MMAP_INTERLEAVED, direct_write_loop }, { NULL, SND_PCM_ACCESS_RW_INTERLEAVED, NULL } };
static void help(void) { int k; printf( "Usage: pcm [OPTION]... [FILE]...\n" "-h,--help help\n" "-D,--device playback device\n" "-r,--rate stream rate in Hz\n" "-c,--channels count of channels in stream\n" "-f,--frequency sine wave frequency in Hz\n" "-b,--buffer ring buffer size in us\n" "-p,--period period size in us\n" "-m,--method transfer method\n" "-o,--format sample format\n" "-v,--verbose show the PCM setup parameters\n" "-n,--noresample do not resample\n" "-e,--pevent enable poll event after each period\n" "\n"); printf("Recognized sample formats are:"); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { const char *s = snd_pcm_format_name(k); if (s) printf(" %s", s); } printf("\n"); printf("Recognized transfer methods are:"); for (k = 0; transfer_methods[k].name; k++) printf(" %s", transfer_methods[k].name); printf("\n"); }
int main (int argc, char *argv[]) { struct option long_option[] = { {"help", 0, NULL, 'h'}, {"device", 1, NULL, 'd'}, {"rate", 1, NULL, 'r'}, {"channels", 1, NULL, 'c'}, {"frequency", 1, NULL, 'f'}, {"buffer", 1, NULL, 'b'}, {"period", 1, NULL, 'p'}, {"method", 1, NULL, 'm'}, {"format", 1, NULL, 'o'}, {"verbose", 1, NULL, 'v'}, {"noresample", 1, NULL, 'n'}, {"pevent", 1, NULL, 'e'}, {NULL, 0, NULL, 0}, };
snd_pcm_t *handle; //ponteiro para pcm handle int err, morehelp; snd_pcm_hw_params_t *hwparams; //parametros de hardware snd_pcm_sw_params_t *swparams; //parametros de software int method = 0; signed short *samples; unsigned int chn; snd_pcm_channel_area_t *areas;
//snd_pcm_hw_params_alloca(&hwparams); //snd_pcm_sw_params_alloca(&swparams);
morehelp = 0; while (1) { int c; if ((c = getopt_long(argc, argv, "hd:r:c:f:b:p:m:o:vne", long_option, NULL)) < 0) break; switch(c) { case 'h': morehelp++; break; case 'd': device = strdup(optarg); break; case 'r': rate = atoi(optarg); rate = rate < 4000 ? 4000 : rate; rate = rate > 196000 ? 19600 : rate; break; case 'c': channels = atoi(optarg); channels = channels < 1 ? 1 : channels; channels = channels > 1024 ? 1024 : channels; break; case 'f': freq = atoi(optarg); freq = freq < 50 ? 50 : freq; freq = freq > 5000 ? 5000 : freq; break; case 'b': buffer_time = atoi(optarg); buffer_time = buffer_time < 1000 ? 1000 : buffer_time; buffer_time = buffer_time > 1000000 ? 1000000 : buffer_time; break; case 'p': period_time = atoi(optarg); period_time = period_time < 1000 ? 1000 : period_time; period_time = period_time > 1000000 ? 1000000 : period_time; break; case 'm': for (method = 0; transfer_methods[method].name; method++) if (!strcasecmp(transfer_methods[method].name, optarg)) break; if (transfer_methods[method].name == NULL) method = 0; break; case 'o': for (format = 0; format < SND_PCM_FORMAT_LAST; format++) { const char *format_name = snd_pcm_format_name(format); if (format_name) if (!strcasecmp(format_name, optarg)) break; } if (format == SND_PCM_FORMAT_LAST) format = SND_PCM_FORMAT_S16; break; case 'v': verbose = 1; break; case 'n': resample = 0; break; case 'e': period_event = 1; break; }//switch }//while
if(morehelp) { help(); return 0; }
//cria um novo output objeto err = snd_output_stdio_attach(&output, &stdout, 0); if (err < 0) { printf("Output failed: %s\n", snd_strerror(err)); return 0; }
printf("Playback device is %s\n", device); printf("Stream parameters are %iHz, %s, %i channels\n", rate, snd_pcm_format_name(format), channels); printf("sine wave rate is %i %.4fHz\n", freq); printf("Using transfer method: %s\n", transfer_methods[method].name);
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); return 0; }
if ((err = set_hwparams(handle, hwparams, transfer_methods[method].access)) < 0) { printf("Setting of hwparams failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if ((err = set_swparams(handle, swparams)) < 0) { printf("Setting of swparams failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); }
if (verbose > 0) snd_pcm_dump(handle, output);
samples = malloc((period_size * channels * snd_pcm_format_physical_width(format)) / 8); if (samples == NULL) { printf("No enough memory\n"); exit(EXIT_FAILURE); }
areas = calloc(channels, sizeof(snd_pcm_channel_area_t)); if (areas == NULL) { printf("No enough memory\n"); exit(EXIT_FAILURE); } for (chn = 0; chn < channels; chn++) { areas[chn].addr = samples; areas[chn].first = chn * snd_pcm_format_physical_width(format); areas[chn].step = channels * snd_pcm_format_physical_width(format); }
err = transfer_methods[method].transfer_loop(handle, samples, areas); if (err < 0) printf("Transfer failed: %s\n", snd_strerror(err));
free(areas); free(samples); snd_pcm_close(handle); return 0; }
Any ideas of what might be happening?? I've been looking for a reasonable explanation but i don't find any related error o online.
Tks in advanced!
Tks!
-------------------
Guilherme Longo Dept. Eng. da Computação Unaerp
Linux User - #484927
*Before Asking http://www.istf.com.br/?page=perguntas
!- I'd rather die on my feet than live on my knees -!
Takashi Iwai wrote:
At Fri, 19 Jun 2009 02:50:35 +0300, Maxim Levitsky wrote:
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Because it harms than helps more often. I've got tons of bug reports and been upset just because users blindly set "Analog Loopback" mixer switch and lost their sound output. This is no mixer element that behaves intuitively -- it shuts out PCM output instead of adding the analog loopback like other hardwares.
The feature is still available, though. You just need to give the driver an instruction via sysfs. Namely, write to sysfs files corresponding to the IDT codec like # echo "loopback yes" > /sys/class/sound/hwC0D0/hints # echo 1 > /sys/class/sound/hwC0D0/reconfig Then the driver will reconfigure with the loopback mixer element.
I have a patch series to add this with a "patch" firmware file to make it easier, which I'm going to send an RFC to this ML soon later. Take a look at topic/hda-patch branch on sound-unstable tree.
thanks,
Takashi _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Thanks Takashi. Its solved the problem.
Tks!
-------------------
Guilherme Longo Dept. Eng. da Computação Unaerp
Linux User - #484927
*Before Asking http://www.istf.com.br/?page=perguntas
!- I'd rather die on my feet than live on my knees -!
Takashi Iwai wrote:
At Fri, 19 Jun 2009 04:27:58 -0300, Guilherme wrote:
//snd_pcm_hw_params_alloca(&hwparams); //snd_pcm_sw_params_alloca(&swparams);
Uncomment these.
Takashi
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 02:50:35 +0300, Maxim Levitsky wrote:
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Because it harms than helps more often. I've got tons of bug reports and been upset just because users blindly set "Analog Loopback" mixer switch and lost their sound output. This is no mixer element that behaves intuitively -- it shuts out PCM output instead of adding the analog loopback like other hardwares.
Here it doesn't shut PCM. I just hear input in headphones, mixed with output. I suspect that this feature might not work correctly on newer devices that ones I have added.
The feature is still available, though. You just need to give the driver an instruction via sysfs. Namely, write to sysfs files corresponding to the IDT codec like # echo "loopback yes" > /sys/class/sound/hwC0D0/hints # echo 1 > /sys/class/sound/hwC0D0/reconfig Then the driver will reconfigure with the loopback mixer element.
I have a patch series to add this with a "patch" firmware file to make it easier, which I'm going to send an RFC to this ML soon later. Take a look at topic/hda-patch branch on sound-unstable tree.
thanks,
Takashi
At Fri, 19 Jun 2009 11:21:41 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 02:50:35 +0300, Maxim Levitsky wrote:
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Because it harms than helps more often. I've got tons of bug reports and been upset just because users blindly set "Analog Loopback" mixer switch and lost their sound output. This is no mixer element that behaves intuitively -- it shuts out PCM output instead of adding the analog loopback like other hardwares.
Here it doesn't shut PCM. I just hear input in headphones, mixed with output. I suspect that this feature might not work correctly on newer devices that ones I have added.
Possibly...
Takashi
On Fri, 2009-06-19 at 10:23 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 11:21:41 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 02:50:35 +0300, Maxim Levitsky wrote:
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Because it harms than helps more often. I've got tons of bug reports and been upset just because users blindly set "Analog Loopback" mixer switch and lost their sound output. This is no mixer element that behaves intuitively -- it shuts out PCM output instead of adding the analog loopback like other hardwares.
Here it doesn't shut PCM. I just hear input in headphones, mixed with output. I suspect that this feature might not work correctly on newer devices that ones I have added.
Possibly...
Probably IDT removed the analog mixer in quest for better quality.
I think it should work fine for "STAC..." parts.
Anyway lets put it back at least for my STAC9227, OK?
Best regards, Maxim Levitsky
At Fri, 19 Jun 2009 14:36:39 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 10:23 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 11:21:41 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 02:50:35 +0300, Maxim Levitsky wrote:
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Because it harms than helps more often. I've got tons of bug reports and been upset just because users blindly set "Analog Loopback" mixer switch and lost their sound output. This is no mixer element that behaves intuitively -- it shuts out PCM output instead of adding the analog loopback like other hardwares.
Here it doesn't shut PCM. I just hear input in headphones, mixed with output. I suspect that this feature might not work correctly on newer devices that ones I have added.
Possibly...
Probably IDT removed the analog mixer in quest for better quality.
I think it should work fine for "STAC..." parts.
Anyway lets put it back at least for my STAC9227, OK?
You can still use it just by adjusting the sysfs (and in near future via a patch file), so I'm not going to take it back for the default.
Takashi
On Fri, 2009-06-19 at 14:50 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 14:36:39 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 10:23 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 11:21:41 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 02:50:35 +0300, Maxim Levitsky wrote:
Hi,
Due to lack of time, I didn't update kernel on my desktop system. Now I notice that analog loopback go removed. Why?
Because it harms than helps more often. I've got tons of bug reports and been upset just because users blindly set "Analog Loopback" mixer switch and lost their sound output. This is no mixer element that behaves intuitively -- it shuts out PCM output instead of adding the analog loopback like other hardwares.
Here it doesn't shut PCM. I just hear input in headphones, mixed with output. I suspect that this feature might not work correctly on newer devices that ones I have added.
Possibly...
Probably IDT removed the analog mixer in quest for better quality.
I think it should work fine for "STAC..." parts.
Anyway lets put it back at least for my STAC9227, OK?
You can still use it just by adjusting the sysfs (and in near future via a patch file), so I'm not going to take it back for the default.
Thanks....
At Fri, 19 Jun 2009 21:18:49 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
echo "loopback yes" > /sys/class/sound/hwC0D0/hints
This is unmerged feature?
root@MAIN:/home/maxim# cat /sys/class/sound/hwC0D2/ dev device/ power/ subsystem/ uevent
I use -git of the kernel
Set CONFIG_SND_HDA_RECONFIG=y.
Takashi
On Fri, 2009-06-19 at 20:44 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 21:18:49 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
echo "loopback yes" > /sys/class/sound/hwC0D0/hints
This is unmerged feature?
root@MAIN:/home/maxim# cat /sys/class/sound/hwC0D2/ dev device/ power/ subsystem/ uevent
I use -git of the kernel
Set CONFIG_SND_HDA_RECONFIG=y.
Takashi
Ok, I see it now, but then I put these commands in /etc/rc.local
echo "loopback=yes" > /sys/class/sound/hwC0D2/hints echo 1 > /sys/class/sound/hwC0D2/reconfig
and I get this, and almost empty mixer
[ 57.353168] hda-codec: reconfiguring [ 57.426105] stac92xx_auto_fill_dac_nids: No available DAC for pin 0xd [ 59.121418] e1000e 0000:00:19.0: irq 29 for MSI/MSI-X [ 59.172115] e1000e 0000:00:19.0: irq 29 for MSI/MSI-X [ 60.585005] e1000e: eth0 NIC Link is Up 100 Mbps Full Duplex, Flow Control: RX/TX [ 60.585011] 0000:00:19.0: eth0: 10/100 speed: disabling TSO
I had to reload the driver and apply the above commands to make it and loopback work...
Why? It all did work fine before.
Regards, Maxim Levitsky
At Sat, 20 Jun 2009 00:05:01 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 20:44 +0200, Takashi Iwai wrote:
At Fri, 19 Jun 2009 21:18:49 +0300, Maxim Levitsky wrote:
On Fri, 2009-06-19 at 08:32 +0200, Takashi Iwai wrote:
echo "loopback yes" > /sys/class/sound/hwC0D0/hints
This is unmerged feature?
root@MAIN:/home/maxim# cat /sys/class/sound/hwC0D2/ dev device/ power/ subsystem/ uevent
I use -git of the kernel
Set CONFIG_SND_HDA_RECONFIG=y.
Takashi
Ok, I see it now, but then I put these commands in /etc/rc.local
echo "loopback=yes" > /sys/class/sound/hwC0D2/hints echo 1 > /sys/class/sound/hwC0D2/reconfig
and I get this, and almost empty mixer
[ 57.353168] hda-codec: reconfiguring [ 57.426105] stac92xx_auto_fill_dac_nids: No available DAC for pin 0xd
Yeah, something is wrong here. Could you run alsa-info.sh with --no-upload option at this stage, and attach the generated file?
thanks,
Takashi
participants (3)
-
Guilherme
-
Maxim Levitsky
-
Takashi Iwai