June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Playback open error: Device or resource busy
by Guilherme 17 Jun '09

17 Jun '09

Mates... I am running a simple app that I found called pcm.c and I can't get through with that. It creates and send a simple sine wave to the speakers! When I run the app I get the following error: guilherme@lap:/home/guilherme/Experimentos# ./PCM Playback device is plughw:0,0 Stream parameters are 44100Hz, S16_LE, 1 channels sine wave rate is 0 0.0000Hz Using transfer method: write Playback open error: Device or resource busy Every application I start playbacks the sound without a problem. Is there any special conf I should do to free the Device? Here 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 = "plughw:0,0"; //playback device aplay -l static snd_pcm_format_t format = SND_PCM_FORMAT_S16; //sample format unsigned 16 bit endian static unsigned int rate = 44100; //stream rate static unsigned int channels = 1; //contagem dos canais static unsigned int buffer_time = 500000; //ring buffer length on us static unsigned int period_time = 100000; //period time in us static double freq = 400; // 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; } Thanks in advance; -- 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 -!

2 2

[alsa-devel] kerneloops.org report for the week of June 14 2009
by Arjan van de Ven 16 Jun '09

16 Jun '09

Hi, First, a word of thanks to the kernel.org team for helping to host the kerneloops.org database on a real machine rather than a small virtualized host. Performance and continuity will be a lot better in the new setup! (as a result of the transition and hosting issues on the original server, there was a downtime of two days, meaning that the total number of reports is lower than typical) New sightings this report include the XFI driver, the i915 gem code and the DMAR code. This week, a total of 4026 oopses and warnings have been reported for kernels 2.6.29 and later. Reports for kernels earlier than 2.6.29 have been omitted from this report. Per file statistics 830 drivers/gpu/drm/i915/i915_gem_tiling.c 470 drivers/net/wireless/iwlwifi/iwl-sta.c 303 kernel/time.c 276 drivers/parport/procfs.c 269 include/linux/slob_def.h 187 include/linux/hrtimer.h 138 arch/x86/kernel/cpu/mtrr/generic.c 120 drivers/pci/dmar.c 65 net/mac80211/rc80211_minstrel.c 62 drivers/net/r8169.c 37 net/ipv4/tcp.c 34 fs/cifs/cifssmb.c Rank 1: i915_gem_set_tiling (warning) Reported 830 times (3700 total reports) [gem] Failure in the tiling code This warning was last seen in version 2.6.30, and first seen in 2.6.29-rc2. More info: http://www.kerneloops.org/searchweek.php?search=i915_gem_set_tiling Rank 2: iwl_set_dynamic_key (warning) Reported 470 times (13682 total reports) "no space for new kew" message. This warning was last seen in version 2.6.30, and first seen in 2.6.27. More info: http://www.kerneloops.org/searchweek.php?search=iwl_set_dynamic_key Rank 3: getnstimeofday (warning) Reported 309 times (2446 total reports) [suspend resume] getnstimeofday() is called before timekeeping is resumed This warning was last seen in version 2.6.29.4, and first seen in 2.6.24. More info: http://www.kerneloops.org/searchweek.php?search=getnstimeofday Rank 4: ct_vm_map (warning) - sleeping in atomic context Reported 285 times (960 total reports) Bug in the creative XFI driver (backported to Fedora) This warning was last seen in version 2.6.29.4, and first seen in 2.6.29-rc4-git1. More info: http://www.kerneloops.org/searchweek.php?search=ct_vm_map Rank 5: parport_device_proc_unregister (warning) Reported 278 times (1810 total reports) Alan has a fix for this on LKML This warning was last seen in version 2.6.29.4, and first seen in 2.6.27. More info: http://www.kerneloops.org/searchweek.php?search=parport_device_proc_unregis… Rank 6: hres_timers_resume (warning) Reported 188 times (1024 total reports) [suspend resume] hres_timers_resume() is incorrectly called with interrupts on This warning was last seen in version 2.6.30-rc6, and first seen in 2.6.24.7. More info: http://www.kerneloops.org/searchweek.php?search=hres_timers_resume Rank 7: generic_get_mtrr (warning) Reported 139 times (1009 total reports) BIOS bug where the MTRRs are not set up correctly This warning was last seen in version 2.6.30, and first seen in 2.6.25.3. More info: http://www.kerneloops.org/searchweek.php?search=generic_get_mtrr Rank 8: dmar_table_init (warning) Reported 121 times (507 total reports) BIOS bug exposed via a WARN_ON This warning was last seen in version 2.6.30-rc8, and first seen in 2.6.29. More info: http://www.kerneloops.org/searchweek.php?search=dmar_table_init Rank 9: minstrel_get_rate (warning) Reported 64 times (384 total reports) Issue with the ath5k driver This warning was last seen in version 2.6.30-rc3, and first seen in 2.6.27.12. More info: http://www.kerneloops.org/searchweek.php?search=minstrel_get_rate Rank 10: dev_watchdog(r8169) (warning) Reported 62 times (9116 total reports) This warning was last seen in version 2.6.30-rc7, and first seen in 2.6.26.6. More info: http://www.kerneloops.org/searchweek.php?search=dev_watchdog(r8169) Rank 11: nv_tx_done_optimized (warning) Reported 61 times (2974 total reports) the NV ethernet driver is getting caught with the new DMA API checks This warning was last seen in version 2.6.30, and first seen in 2.6.29-rc3-git7. More info: http://www.kerneloops.org/searchweek.php?search=nv_tx_done_optimized Rank 12: tcp_recvmsg (warning) Reported 37 times (3797 total reports) fixed in 2.6.30 (commit 775273131810caa41dfc7f9e552ea5d8508caf40) This warning was last seen in version 2.6.30, and first seen in 2.6.25. More info: http://www.kerneloops.org/searchweek.php?search=tcp_recvmsg -- Arjan van de Ven Intel Open Source Technology Centre For development, discussion and tips for power savings, visit http://www.lesswatts.org

3 3

Re: [alsa-devel] ALSA 1.0.20 Speex PCM Plugin
by Takashi Iwai 16 Jun '09

16 Jun '09

At Mon, 8 Jun 2009 15:56:38 -0400, Robert Krakora wrote: > > Takashi: > > I applied your Speex echo cancellation patch. I had to modify it > since the 'buf_out' buffer was not being malloc'd. Attached are the > files affected by your patch. The echo cancellation seems to work > pretty well. However, turning on 'denoise' seems to cause audio to > drop in and out at a very periodic rate. Do you have any example of > how to set up all of the parameters? A tutorial or something? Thanks > again. No idea, I just wrote it up blindly :) Can the preprocessor be dropped, possibly, like the patch below? Takashi --- diff --git a/doc/speexdsp.txt b/doc/speexdsp.txt index 875fc19..5b5e5a0 100644 --- a/doc/speexdsp.txt +++ b/doc/speexdsp.txt @@ -12,7 +12,7 @@ using libspeex DSP API. You can use the plugin with the plugin type Then record like - % arecord -fdat -c1 -Dplug:speex foo.wav + % arecord -fdat -c1 -Dplug:my_pcm foo.wav so that you'll get 48kHz mono stream with the denoising effect. @@ -44,6 +44,15 @@ The following parameters can be set optionally: A boolean value to enable/disable dereverb function. Default is no. +* echo + + A boolean value to enable/disable echo-cancellation function. + Default is no. + +* filter_length + + Number of samples of echo to cancel. As default it's 256. + For example, you can enable agc like pcm.my_pcm { diff --git a/speex/pcm_speex.c b/speex/pcm_speex.c index 7bb9213..757a400 100644 --- a/speex/pcm_speex.c +++ b/speex/pcm_speex.c @@ -1,5 +1,5 @@ /* - * Speex preprocess plugin + * Speex DSP plugin * * Copyright (c) 2009 by Takashi Iwai <tiwai(a)suse.de> * @@ -21,12 +21,15 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_external.h> #include <speex/speex_preprocess.h> +#include <speex/speex_echo.h> -/* preprocessing parameters */ +/* DSP parameters */ struct spx_parms { int frames; int denoise; int agc; + int echo; + int filter_length; float agc_level; int dereverb; float dereverb_decay; @@ -38,7 +41,9 @@ typedef struct { struct spx_parms parms; /* instance and intermedate buffer */ SpeexPreprocessState *state; + SpeexEchoState *echo_state; short *buf; + short *outbuf; /* running states */ unsigned int filled; unsigned int processed; @@ -64,6 +69,18 @@ spx_transfer(snd_pcm_extplug_t *ext, short *src = area_addr(src_areas, src_offset); short *dst = area_addr(dst_areas, dst_offset); unsigned int count = size; + short *databuf; + + if (!spx->state && !spx->echo_state) { + /* no DSP processing */ + memcpy(dst, src, count * 2); + return size; + } + + if (spx->echo_state) + databuf = spx->outbuf; + else + databuf = spx->buf; while (count > 0) { unsigned int chunk; @@ -72,14 +89,20 @@ spx_transfer(snd_pcm_extplug_t *ext, else chunk = count; if (spx->processed) - memcpy(dst, spx->buf + spx->filled, chunk * 2); + memcpy(dst, databuf + spx->filled, chunk * 2); else memset(dst, 0, chunk * 2); dst += chunk; memcpy(spx->buf + spx->filled, src, chunk * 2); spx->filled += chunk; if (spx->filled == spx->parms.frames) { - speex_preprocess_run(spx->state, spx->buf); + if (spx->echo_state) + speex_echo_capture(spx->echo_state, spx->buf, + spx->outbuf); + if (spx->state) + speex_preprocess_run(spx->state, databuf); + if (spx->echo_state) + speex_echo_playback(spx->echo_state, databuf); spx->processed = 1; spx->filled = 0; } @@ -94,6 +117,9 @@ static int spx_init(snd_pcm_extplug_t *ext) { snd_pcm_speex_t *spx = (snd_pcm_speex_t *)ext; + spx->filled = 0; + spx->processed = 0; + if (!spx->buf) { spx->buf = malloc(spx->parms.frames * 2); if (!spx->buf) @@ -101,12 +127,43 @@ static int spx_init(snd_pcm_extplug_t *ext) } memset(spx->buf, 0, spx->parms.frames * 2); - if (spx->state) + if (!spx->outbuf) { + spx->outbuf = malloc(spx->parms.frames * 2); + if (!spx->outbuf) + return -ENOMEM; + } + memset(spx->outbuf, 0, spx->parms.frames * 2); + + if (spx->state) { speex_preprocess_state_destroy(spx->state); + spx->state = NULL; + } + if (spx->echo_state) { + speex_echo_state_destroy(spx->echo_state); + spx->echo_state = NULL; + } + + if (spx->parms.echo) { + spx->echo_state = speex_echo_state_init(spx->parms.frames, + spx->parms.filter_length); + if (!spx->echo_state) + return -EIO; + speex_echo_ctl(spx->echo_state, SPEEX_ECHO_SET_SAMPLING_RATE, + &spx->ext.rate); + } + + /* no preprocessor? */ + if (!spx->parms.denoise && !spx->parms.agc && !spx->parms.dereverb) + return 0; + spx->state = speex_preprocess_state_init(spx->parms.frames, spx->ext.rate); if (!spx->state) return -EIO; + if (spx->echo_state) + speex_preprocess_ctl(spx->state, + SPEEX_PREPROCESS_SET_ECHO_STATE, + spx->echo_state); speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_DENOISE, &spx->parms.denoise); @@ -120,18 +177,18 @@ static int spx_init(snd_pcm_extplug_t *ext) &spx->parms.dereverb_decay); speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_DEREVERB_LEVEL, &spx->parms.dereverb_level); - - spx->filled = 0; - spx->processed = 0; return 0; } static int spx_close(snd_pcm_extplug_t *ext) { snd_pcm_speex_t *spx = (snd_pcm_speex_t *)ext; + free(spx->outbuf); free(spx->buf); if (spx->state) speex_preprocess_state_destroy(spx->state); + if (spx->echo_state) + speex_echo_state_destroy(spx->echo_state); return 0; } @@ -205,6 +262,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex) .dereverb = 0, .dereverb_decay = 0, .dereverb_level = 0, + .echo = 0, + .filter_length = 256, }; snd_config_for_each(i, next, conf) { @@ -242,6 +301,13 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex) &parms.dereverb_level); if (err) goto ok; + err = get_bool_parm(n, id, "echo", &parms.echo); + if (err) + goto ok; + err = get_int_parm(n, id, "filter_length", + &parms.filter_length); + if (err) + goto ok; SNDERR("Unknown field %s", id); err = -EINVAL; ok: @@ -259,7 +325,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex) return -ENOMEM; spx->ext.version = SND_PCM_EXTPLUG_VERSION; - spx->ext.name = "Speex Denoise Plugin"; + spx->ext.name = "Speex DSP Plugin"; spx->ext.callback = &speex_callback; spx->ext.private_data = spx; spx->parms = parms;

2 2

[alsa-devel] [PATCH 2/3] ASoC: pxa-ssp.c, Automatically set TDM when needed
by Daniel Ribeiro 16 Jun '09

16 Jun '09

* Automatically sets TDM mode for frame_width larger than 32 bits, if the user doesn't setup the TDM slots with set_tdm_slot(). * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. * Makes SSCR0_MOD optional. * Clears SSCR1_RWOT case SSCR0_MOD is set. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 92 +++++++++++++++++++++++++++-------------------- 1 files changed, 53 insertions(+), 39 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 5b9ed64..d60492e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -452,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | + SSCR0_ACS | SSCR0_DSS | SSCR0_EDSS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -496,7 +496,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; break; @@ -532,48 +532,70 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sscr1, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int slot_width, frame_width = 0; + + /* check if the user explicitly set a slot_width */ + sscr0 = ssp_read_reg(ssp, SSCR0); + + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) + slot_width = (sscr0 & SSCR0_DSS) + + (sscr0 & SSCR0_EDSS ? 17 : 1); + else + frame_width = slot_width = width * chn; /* generate correct DMA params */ if (cpu_dai->dma_data) kfree(cpu_dai->dma_data); - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), + cpu_dai->dma_data = ssp_get_dma_params(ssp, slot_width > 16, substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (sscr0 & SSCR0_SSE) return 0; - /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; + if (slot_width == 16 && cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; #endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; + + if (frame_width > 0) { + /* Not using network mode */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (frame_width > 32) { + /* + * Network mode is needed to support this frame width. + * We assume the wire is not networked and setup a + * fake network mode here. Use as many slots as needed + * each with 32 bits. + */ + int slots = frame_width / 32; + + sscr0 |= SSCR0_MOD; + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* + * Set active slots. Only set an active TX slot + * if we are going to use it. + */ + ssp_write_reg(ssp, SSRSA, slots - 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssp_write_reg(ssp, SSTSA, slots - 1); + } } + + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ + if (sscr0 & SSCR0_MOD) { + sscr1 = ssp_read_reg(ssp, SSCR1); + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); + } + ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -620,14 +642,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; -- tg: (b977873..) asoc/ssp-internals (depends on: asoc/set_tdm_slot) -- Daniel Ribeiro

1 0

[alsa-devel] [PATCH 1/3] ASoC: magician: fix PXA SSP clock polarity
by Philipp Zabel 15 Jun '09

15 Jun '09

Follow-up fix needed since "ASoC: pxa-ssp.c fix clock/frame invert". Signed-off-by: Philipp Zabel <philipp.zabel(a)gmail.com> --- sound/soc/pxa/magician.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index a04ddc3..fa0e7e6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -183,7 +183,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; -- 1.6.3.1

6 12

[alsa-devel] PROBLEM WITH EMAIL
by Guilherme 15 Jun '09

15 Jun '09

MATES... My emails are not getting through...! How can I send e-mail to the list? -- 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 -!

2 1

[alsa-devel] Timing of linked devices
by louis＠museresearch.com 15 Jun '09

15 Jun '09

I am developing an application which opens both inputs and outputs for digital and s/pdif (so, 4 devices). The sync IDs all match. However, under certain conditions, I am receiving partial buffers from the input. So, I have a few questions: Does linking input and output devices, from a programmer's perspective, mean that input is ready at precisely the same time that the device is ready to output a buffer? Or, does it only mean that the ins and outs will be synchronized for each given buffer? More to the point, will blocking on a write also accurately tell the program when the device is ready for a read (including under non-ideal circumstances like after a bunch of xruns)? And, in the case that I am doing a non-blocking read (pcm_read) from a buffer which is not yet ready, should I receive an outdated buffer, a blank buffer, or one which has both partially new and partially old data? I am using ALSA 1.0.15 and the ICE1724 driver. Thanks in advance for your help! - Louis

2 1

Re: [alsa-devel] snd-ctxfi oops
by The Source 15 Jun '09

15 Jun '09

On 12.06.2009 22:22, The Source wrote: > On 12.06.2009 11:20, Takashi Iwai wrote: >> At Fri, 12 Jun 2009 07:52:22 +0400, >> The Source wrote: >>> Hello. I use snd-ctxfi from 8 June snapshot. I hadn't oopses since I >>> installed this driver but recently I got one again: >>> >>> Kernel failure message 1: >>> BUG: sleeping function called from invalid context at mm/slub.c:1599 >> Could you try the patch below? >> >> >> thanks, >> >> Takashi >> >> --- >> diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c >> index 80fb2ba..b0adc80 100644 >> --- a/sound/pci/ctxfi/ctatc.c >> +++ b/sound/pci/ctxfi/ctatc.c >> @@ -259,7 +259,6 @@ static int atc_pcm_playback_prepare(struct ct_atc >> *atc, struct ct_atc_pcm *apcm) >> int n_amixer = apcm->substream->runtime->channels, i = 0; >> int device = apcm->substream->pcm->device; >> unsigned int pitch; >> - unsigned long flags; >> >> if (NULL != apcm->src) { >> /* Prepared pcm playback */ >> @@ -311,10 +310,10 @@ static int atc_pcm_playback_prepare(struct >> ct_atc *atc, struct ct_atc_pcm *apcm) >> src = apcm->src; >> for (i = 0; i< n_amixer; i++) { >> amixer = apcm->amixers[i]; >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> amixer->ops->setup(amixer,&src->rsc, >> INIT_VOL, atc->pcm[i+device*2]); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> src = src->ops->next_interleave(src); >> if (NULL == src) >> src = apcm->src; >> @@ -865,7 +864,6 @@ static int >> spdif_passthru_playback_setup(struct ct_atc *atc, struct ct_atc_pcm >> *apcm) >> { >> struct dao *dao = container_of(atc->daios[SPDIFOO], struct dao, >> daio); >> - unsigned long flags; >> unsigned int rate = apcm->substream->runtime->rate; >> unsigned int status; >> int err; >> @@ -885,7 +883,7 @@ spdif_passthru_playback_setup(struct ct_atc *atc, >> struct ct_atc_pcm *apcm) >> return -ENOENT; >> } >> >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> dao->ops->get_spos(dao,&status); >> if (((status>> 24)& IEC958_AES3_CON_FS) != iec958_con_fs) { >> status&= ((~IEC958_AES3_CON_FS)<< 24); >> @@ -895,7 +893,7 @@ spdif_passthru_playback_setup(struct ct_atc *atc, >> struct ct_atc_pcm *apcm) >> } >> if ((rate != atc->pll_rate)&& (32000 != rate)) >> err = atc_pll_init(atc, rate); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> >> return err; >> } >> @@ -908,7 +906,6 @@ spdif_passthru_playback_prepare(struct ct_atc >> *atc, struct ct_atc_pcm *apcm) >> struct dao *dao; >> int err; >> int i; >> - unsigned long flags; >> >> if (NULL != apcm->src) >> return 0; >> @@ -934,13 +931,13 @@ spdif_passthru_playback_prepare(struct ct_atc >> *atc, struct ct_atc_pcm *apcm) >> src = apcm->src; >> } >> /* Connect to SPDIFOO */ >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> dao = container_of(atc->daios[SPDIFOO], struct dao, daio); >> amixer = apcm->amixers[0]; >> dao->ops->set_left_input(dao,&amixer->rsc); >> amixer = apcm->amixers[1]; >> dao->ops->set_right_input(dao,&amixer->rsc); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> >> ct_timer_prepare(apcm->timer); >> >> @@ -1088,7 +1085,6 @@ static int atc_spdif_out_set_status(struct >> ct_atc *atc, unsigned int status) >> >> static int atc_spdif_out_passthru(struct ct_atc *atc, unsigned char >> state) >> { >> - unsigned long flags; >> struct dao_desc da_dsc = {0}; >> struct dao *dao; >> int err; >> @@ -1096,7 +1092,7 @@ static int atc_spdif_out_passthru(struct ct_atc >> *atc, unsigned char state) >> struct rsc *rscs[2] = {NULL}; >> unsigned int spos = 0; >> >> - spin_lock_irqsave(&atc->atc_lock, flags); >> + mutex_lock(&atc->atc_mutex); >> dao = container_of(atc->daios[SPDIFOO], struct dao, daio); >> da_dsc.msr = state ? 1 : atc->msr; >> da_dsc.passthru = state ? 1 : 0; >> @@ -1114,7 +1110,7 @@ static int atc_spdif_out_passthru(struct ct_atc >> *atc, unsigned char state) >> } >> dao->ops->set_spos(dao, spos); >> dao->ops->commit_write(dao); >> - spin_unlock_irqrestore(&atc->atc_lock, flags); >> + mutex_unlock(&atc->atc_mutex); >> >> return err; >> } >> @@ -1572,7 +1568,7 @@ int __devinit ct_atc_create(struct snd_card >> *card, struct pci_dev *pci, >> atc->msr = msr; >> atc->chip_type = chip_type; >> >> - spin_lock_init(&atc->atc_lock); >> + mutex_init(&atc->atc_mutex); >> >> /* Find card model */ >> err = atc_identify_card(atc); >> diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h >> index a033472..9fe620e 100644 >> --- a/sound/pci/ctxfi/ctatc.h >> +++ b/sound/pci/ctxfi/ctatc.h >> @@ -19,7 +19,7 @@ >> #define CTATC_H >> >> #include<linux/types.h> >> -#include<linux/spinlock_types.h> >> +#include<linux/mutex.h> >> #include<linux/pci.h> >> #include<linux/timer.h> >> #include<sound/core.h> >> @@ -90,7 +90,7 @@ struct ct_atc { >> void (*unmap_audio_buffer)(struct ct_atc *atc, struct >> ct_atc_pcm *apcm); >> unsigned long (*get_ptp_phys)(struct ct_atc *atc, int index); >> >> - spinlock_t atc_lock; >> + struct mutex atc_mutex; >> >> int (*pcm_playback_prepare)(struct ct_atc *atc, >> struct ct_atc_pcm *apcm); >> > No oopses with this patch so far, thank you. Tested the patch more. Really no oopses. Instead of them I get rare system lockups.

2 2

[alsa-devel] ASoC: About amp driver
by Joonyoung Shim 15 Jun '09

15 Jun '09

Hi. I have one question. Currently, the amp driver of ASoC is included in machine driver. (ex. neo1973_wm8753 machine driver) I want to use the same amp driver in more than one machine driver, but then the same amp driver code exists double in each machine driver. So, how about spliting between the amp driver and the machine driver? and Is it possible?

2 3

[alsa-devel] [PATCH 0/4] ASoC updates for 2.6.31
by Mark Brown 15 Jun '09

15 Jun '09

The following changes since commit 236e6723bedb483b2ebf73ada01e2c853c5cac01: Mark Brown (1): ASoC: Fix lm4857 control are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Mark Brown (3): ASoC: Remove odd bit clock ratios for WM8903 ASoC: Revert duplicated code in SSM2602 driver ASoC: Instantiate any forgotten DAPM widgets Philipp Zabel (1): ASoC: magician: fix PXA SSP clock polarity sound/soc/codecs/ssm2602.c | 4 ---- sound/soc/codecs/wm8903.c | 4 ---- sound/soc/pxa/magician.c | 2 +- sound/soc/soc-core.c | 3 +++ 4 files changed, 4 insertions(+), 9 deletions(-)

2 5

Re: [alsa-devel] [Alsa-user] Problems with Soundblaster X-Fi
by WILMES Andreas 15 Jun '09

15 Jun '09

Daniel Herrmann wrote: > Same error than before... > > Didn't change anything, but thanks Are you sure that you're using the same gcc-compiler on the alsa modules as same as on the kernel? The compiler-versions might being different.

1 0

Re: [alsa-devel] [Alsa-user] Problems with Soundblaster X-Fi
by WILMES Andreas 15 Jun '09

15 Jun '09

Daniel Herrmann wrote: > WILMES Andreas schrieb: >> Daniel Herrmann wrote: >>> WILMES Andreas schrieb: >>>> Daniel Herrmann wrote: >>>> >>>>> Hey all, >>>>> >>>>> i tried out the new devel-snapshot including alsa-support for my >>>>> Soundblaster x-Fi card. >>>>> >>>>> Installed it via ./configure; make; make install and rebooted. >>>>> >>>>> The driver didnt load at all. >>>>> >>>>> Trying to load the driver manually brings up the following message: >>>>> >>>>> >>>>>> modprobe snd-ctxfi >>>>>> >>>>> FATAL: Error inserting snd_ctxfi >>>>> (/lib/modules/2.6.28-11-generic/kernel/sound/pci/ctxfi/snd-ctxfi.ko): >>>>> Unknown symbol in module, or unknown parameter >>>>> >>>>> >>>>>> dmesg shows the following: >>>>>> >>>>> [ 2146.563073] snd_ctxfi: disagrees about version of symbol >>>>> snd_ctl_add >>>>> [ 2146.563081] snd_ctxfi: Unknown symbol snd_ctl_add >>>>> [ 2146.563291] snd_ctxfi: disagrees about version of symbol >>>>> snd_pcm_new >>>>> [ 2146.563295] snd_ctxfi: Unknown symbol snd_pcm_new >>>>> [ 2146.563482] snd_ctxfi: disagrees about version of symbol >>>>> snd_card_register >>>>> >>>>> [...] and so on, about 60 lines >>>>> >>>>> [ 3198.031060] snd_ctxfi: disagrees about version of symbol >>>>> snd_pcm_period_elapsed >>>>> [ 3198.031063] snd_ctxfi: Unknown symbol snd_pcm_period_elapsed >>>>> >>>>> >>>>>> lspci | grep -i audio >>>>>> >>>>> 00:0a.0 Multimedia audio controller: Creative Labs SB X-Fi >>>>> >>>>> Currently using Ubuntu 9.04 Jaunty with the current >>>>> 2.6.28-11-generic kernel >>>>> >>>>> Maybe anyone has an idea, why i cant install the driver? >>>>> It worked for me before, but unfortunatly i had to reinstall >>>>> ubuntu after getting a weird dependeny problem in apt. >>>>> Not shure why it doesn't do so now... >>>>> >>>>> Thanks in advance for your help and thanks for the great work you do! >>>>> >>>>> >>>> I had the same error. Then I rebooted and the error disappeared. >>>> Hope that will help. >>>> >>> Hm, i rebooted more than once - same error. >>> >>> Daniel >>> >> I deleted /usr/include/asm and I symlinked it to >> /usr/src/linux-headers-2.6.28-11/include/asm-generic and I did a >> recompile. After rebooting then the error disappeared. >> > HM... You deleted it and made a symlink from it? How does this work? > How can i link something deleted?! > > Can't follow you ;) Oh I'm Sorry. /usr/include/asm was a static folder containing a lot of files before. I linked /usr/src/linux-headers-2.6.28-11/include/asm-generic to /usr/include/asm then.

1 0

[alsa-devel] [PATCH v4 3/4] ARM: DaVinci: ASoC: Add mcasp support for DM646x
by Chaithrika U S 15 Jun '09

15 Jun '09

Adds driver support for the two instances of McASP on TI's DM646x. The multichannel audio serial port (McASP) functions as a general-purpose audio serial port optimized for the needs of multichannel audio application. (http://www.ti.com/litv/pdf/spruer1b). There are two instances of McASP on DM646x. The McASP0 module includes up to 4 serializers that can be individually enabled to either transmit or receive in different modes. The McASP1 module is limited with only 1 pinned-out serializer that can be enabled to only transmit in DIT mode (neither receiving in any mode nor transmitting in either Burst or TDM mode is supported). McASP0 consists of transmit and receive sections that may operate synchronized, or completely independently with separate master clocks, bit clocks, and frame syncs, and using different transmit modes with different bit-stream formats. Signed-off-by: Steve Chen <schen(a)mvista.com> Signed-off-by: Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies against ALSA mainline - the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/Kconfig | 13 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-mcasp.c | 874 +++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 55 +++ sound/soc/davinci/davinci-pcm.h | 18 +- 5 files changed, 955 insertions(+), 8 deletions(-) create mode 100644 sound/soc/davinci/davinci-mcasp.c create mode 100644 sound/soc/davinci/davinci-mcasp.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 411a710..6802dd5 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" depends on SND_DAVINCI_SOC @@ -19,6 +22,16 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci DM6446 or DM355 EVM platforms. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_SPDIF + + help + Say Y if you want to add support for SoC audio on TI + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1..67be54f 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,16 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-mcasp-objs:= davinci-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c new file mode 100644 index 0000000..b27aab6 --- /dev/null +++ b/sound/soc/davinci/davinci-mcasp.c @@ -0,0 +1,874 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * Multi-channel Audio Serial Port Driver + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left(even TDM Slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right(odd TDM slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left(even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right(odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD (val)(val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static int davinci_mcasp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); +} + +static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct davinci_audio_dev *dev, + int channel_size) +{ + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + break; + + default: + return -EINVAL; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + return 0; +} + +static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +{ + int i; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + else if (dev->serial_dir[i] == RX_MODE) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + } +} + +static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) +{ + int i, active_slots; + u32 mask = 0; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + } else { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct davinci_audio_dev *dev) +{ + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + + davinci_hw_common_param(dev, substream->stream); + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(dev); + else + davinci_hw_param(dev, substream->stream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + + default: + printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); + return -EINVAL; + } + davinci_config_channel_size(dev, word_length); + + return 0; +} + +static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(dev, substream->stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(dev, substream->stream); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { + .startup = davinci_mcasp_startup, + .trigger = davinci_mcasp_trigger, + .hw_params = davinci_mcasp_hw_params, + .set_fmt = davinci_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_mcasp_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_mcasp_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_mcasp_dai); + +static int davinci_mcasp_probe(struct platform_device *pdev) +{ + struct davinci_pcm_dma_params *dma_data; + struct resource *mem, *ioarea, *res; + struct snd_platform_data *pdata; + struct davinci_audio_dev *dev; + int count = 0; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2, + GFP_KERNEL); + if (!dma_data) { + ret = -ENOMEM; + goto err_release_dev; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + goto err_release_data; + } + + pdata = pdev->dev.platform_data; + dev->clk = clk_get(&pdev->dev, pdata->clk_name); + if (IS_ERR(dev->clk)) { + ret = -ENODEV; + goto err_release_region; + } + + clk_enable(dev->clk); + + dev->base = (void __iomem *)IO_ADDRESS(mem->start); + dev->op_mode = pdata->op_mode; + dev->tdm_slots = pdata->tdm_slots; + dev->num_serializer = pdata->num_serializer; + dev->serial_dir = pdata->serial_dir; + dev->codec_fmt = pdata->codec_fmt; + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev->base)); + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count]; + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev->base)); + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count]; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + davinci_mcasp_dai[pdev->id].private_data = dev; + davinci_mcasp_dai[pdev->id].dev = &pdev->dev; + ret = snd_soc_register_dai(&davinci_mcasp_dai[pdev->id]); + + if (ret != 0) + goto err_release_region; + return 0; + +err_release_region: + release_mem_region(mem->start, (mem->end - mem->start) + 1); +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); + + return ret; +} + +static int davinci_mcasp_remove(struct platform_device *pdev) +{ + struct davinci_pcm_dma_params *dma_data; + struct davinci_audio_dev *dev; + struct resource *mem; + + snd_soc_unregister_dai(&davinci_mcasp_dai[pdev->id]); + dev = davinci_mcasp_dai[pdev->id].private_data; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + + dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(dma_data); + kfree(dev); + + return 0; +} + +static struct platform_driver davinci_mcasp_driver = { + .probe = davinci_mcasp_probe, + .remove = davinci_mcasp_remove, + .driver = { + .name = "davinci-mcasp", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_mcasp_init(void) +{ + return platform_driver_register(&davinci_mcasp_driver); +} +module_init(davinci_mcasp_init); + +static void __exit davinci_mcasp_exit(void) +{ + platform_driver_unregister(&davinci_mcasp_driver); +} +module_exit(davinci_mcasp_exit); + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h new file mode 100644 index 0000000..36b7104 --- /dev/null +++ b/sound/soc/davinci/davinci-mcasp.h @@ -0,0 +1,55 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * MCASP related definitions + * + * Author: Nirmal Pandey <n-pandey(a)ti.com>, + * Suresh Rajashekara <suresh.r(a)ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source(a)mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef DAVINCI_MCASP_H +#define DAVINCI_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_mcasp_dai[]; + +#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000 +#define DAVINCI_MCASP_I2S_DAI 0 +#define DAVINCI_MCASP_DIT_DAI 1 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..eb4287f 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,19 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + enum dma_event_q eventq_no; /* event queue number */ + unsigned char data_type; /* xfer data type */ + unsigned char convert_mono_stereo; }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

3 2

Re: [alsa-devel] [Alsa-user] Problems with Soundblaster X-Fi
by WILMES Andreas 15 Jun '09

15 Jun '09

Daniel Herrmann wrote: > WILMES Andreas schrieb: >> Daniel Herrmann wrote: >> >>> Hey all, >>> >>> i tried out the new devel-snapshot including alsa-support for my >>> Soundblaster x-Fi card. >>> >>> Installed it via ./configure; make; make install and rebooted. >>> >>> The driver didnt load at all. >>> >>> Trying to load the driver manually brings up the following message: >>> >>> >>>> modprobe snd-ctxfi >>>> >>> FATAL: Error inserting snd_ctxfi >>> (/lib/modules/2.6.28-11-generic/kernel/sound/pci/ctxfi/snd-ctxfi.ko): >>> Unknown symbol in module, or unknown parameter >>> >>> >>>> dmesg shows the following: >>>> >>> [ 2146.563073] snd_ctxfi: disagrees about version of symbol snd_ctl_add >>> [ 2146.563081] snd_ctxfi: Unknown symbol snd_ctl_add >>> [ 2146.563291] snd_ctxfi: disagrees about version of symbol snd_pcm_new >>> [ 2146.563295] snd_ctxfi: Unknown symbol snd_pcm_new >>> [ 2146.563482] snd_ctxfi: disagrees about version of symbol >>> snd_card_register >>> >>> [...] and so on, about 60 lines >>> >>> [ 3198.031060] snd_ctxfi: disagrees about version of symbol >>> snd_pcm_period_elapsed >>> [ 3198.031063] snd_ctxfi: Unknown symbol snd_pcm_period_elapsed >>> >>> >>>> lspci | grep -i audio >>>> >>> 00:0a.0 Multimedia audio controller: Creative Labs SB X-Fi >>> >>> Currently using Ubuntu 9.04 Jaunty with the current >>> 2.6.28-11-generic kernel >>> >>> Maybe anyone has an idea, why i cant install the driver? >>> It worked for me before, but unfortunatly i had to reinstall ubuntu >>> after getting a weird dependeny problem in apt. >>> Not shure why it doesn't do so now... >>> >>> Thanks in advance for your help and thanks for the great work you do! >>> >>> >> I had the same error. Then I rebooted and the error disappeared. Hope >> that will help. >> > Hm, i rebooted more than once - same error. > > Daniel > I deleted /usr/include/asm and I symlinked it to /usr/src/linux-headers-2.6.28-11/include/asm-generic and I did a recompile. After rebooting then the error disappeared.

1 0

[alsa-devel] [PATCH v4 1/4] ARM: DaVinci: ASoC: Add the platform devices for ASP
by Chaithrika U S 15 Jun '09

15 Jun '09

1) Registers the platform devices for ASP on dm355, dm644x and dm646x so that the machine driver can probe to get ASP related platform data. 2) Move towards definition of the asp clocks using physical name(for dm355 and dm644x) 3) Add platform data to board specific files. Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git arch/arm/mach-davinci/board-dm355-evm.c | 7 +++ arch/arm/mach-davinci/board-dm644x-evm.c | 5 ++ arch/arm/mach-davinci/board-dm646x-evm.c | 33 ++++++++++++ arch/arm/mach-davinci/dm355.c | 43 +++++++++++++++- arch/arm/mach-davinci/dm644x.c | 36 +++++++++++++- arch/arm/mach-davinci/dm646x.c | 74 +++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 35 ++++++++++++- arch/arm/mach-davinci/include/mach/dm355.h | 5 ++ arch/arm/mach-davinci/include/mach/dm644x.h | 2 + arch/arm/mach-davinci/include/mach/dm646x.h | 3 + 10 files changed, 238 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm355-evm.c b/arch/arm/mach-davinci/board-dm355-evm.c index 5ac2f56..78a9604 100644 --- a/arch/arm/mach-davinci/board-dm355-evm.c +++ b/arch/arm/mach-davinci/board-dm355-evm.c @@ -118,6 +118,10 @@ static struct davinci_i2c_platform_data i2c_pdata = { .bus_delay = 0 /* usec */, }; +static struct snd_platform_data dm355_evm_snd_data = { + .clk_name = "asp1", +}; + static int dm355evm_mmc_gpios = -EINVAL; static void dm355evm_mmcsd_gpios(unsigned gpio) @@ -280,6 +284,9 @@ static __init void dm355_evm_init(void) dm355_init_spi0(BIT(0), dm355_evm_spi_info, ARRAY_SIZE(dm355_evm_spi_info)); + + /* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ + dm355_init_asp1(ASP1_TX_EVT_EN | ASP1_RX_EVT_EN, &dm355_evm_snd_data); } static __init void dm355_evm_irq_init(void) diff --git a/arch/arm/mach-davinci/board-dm644x-evm.c b/arch/arm/mach-davinci/board-dm644x-evm.c index d9d4045..30b2fd4 100644 --- a/arch/arm/mach-davinci/board-dm644x-evm.c +++ b/arch/arm/mach-davinci/board-dm644x-evm.c @@ -226,6 +226,10 @@ static struct platform_device ide_dev = { }, }; +static struct snd_platform_data dm644x_evm_snd_data = { + .clk_name = "asp0", +}; + /*----------------------------------------------------------------------*/ /* @@ -667,6 +671,7 @@ static __init void davinci_evm_init(void) davinci_setup_mmc(0, &dm6446evm_mmc_config); davinci_serial_init(&uart_config); + dm644x_init_asp(&dm644x_evm_snd_data); soc_info->emac_pdata->phy_mask = DM644X_EVM_PHY_MASK; soc_info->emac_pdata->mdio_max_freq = DM644X_EVM_MDIO_FREQUENCY; diff --git a/arch/arm/mach-davinci/board-dm646x-evm.c b/arch/arm/mach-davinci/board-dm646x-evm.c index e17de63..575c6ca 100644 --- a/arch/arm/mach-davinci/board-dm646x-evm.c +++ b/arch/arm/mach-davinci/board-dm646x-evm.c @@ -207,6 +207,37 @@ static struct at24_platform_data eeprom_info = { .context = (void *)0x7f00, }; +static u8 dm646x_iis_serializer_direction[] = { + TX_MODE, RX_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static u8 dm646x_dit_serializer_direction[] = { + TX_MODE, +}; + +static struct snd_platform_data dm646x_evm_snd_data[] = { + { + .clk_name = "mcasp0", + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(dm646x_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = dm646x_iis_serializer_direction, + .eventq_no = EVENTQ_0, + }, + { + .clk_name = "mcasp1", + .tx_dma_offset = 0x400, + .rx_dma_offset = 0, + .op_mode = DAVINCI_MCASP_DIT_MODE, + .num_serializer = ARRAY_SIZE(dm646x_dit_serializer_direction), + .tdm_slots = 32, + .serial_dir = dm646x_dit_serializer_direction, + .eventq_no = EVENTQ_0, + }, +}; + static struct i2c_board_info __initdata i2c_info[] = { { I2C_BOARD_INFO("24c256", 0x50), @@ -240,6 +271,8 @@ static __init void evm_init(void) evm_init_i2c(); davinci_serial_init(&uart_config); + dm646x_init_mcasp0(&dm646x_evm_snd_data[0]); + dm646x_init_mcasp1(&dm646x_evm_snd_data[1]); soc_info->emac_pdata->phy_mask = DM646X_EVM_PHY_MASK; soc_info->emac_pdata->mdio_max_freq = DM646X_EVM_MDIO_FREQUENCY; diff --git a/arch/arm/mach-davinci/dm355.c b/arch/arm/mach-davinci/dm355.c index 9baeed3..1298d8d 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -30,6 +30,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -360,8 +361,8 @@ static struct davinci_clk dm355_clks[] = { CLK(NULL, "uart1", &uart1_clk), CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), - CLK("soc-audio.0", NULL, &asp0_clk), - CLK("soc-audio.1", NULL, &asp1_clk), + CLK(NULL, "asp0", &asp0_clk), + CLK(NULL, "asp1", &asp1_clk), CLK("davinci_mmc.0", NULL, &mmcsd0_clk), CLK("davinci_mmc.1", NULL, &mmcsd1_clk), CLK(NULL, "spi0", &spi0_clk), @@ -623,6 +624,31 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +static struct resource dm355_asp1_resources[] = { + { + .start = DAVINCI_ASP1_BASE, + .end = DAVINCI_ASP1_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP1_TX, + .end = DAVINCI_DMA_ASP1_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP1_RX, + .end = DAVINCI_DMA_ASP1_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm355_asp1_device = { + .name = "davinci-asp", + .id = -1, + .num_resources = ARRAY_SIZE(dm355_asp1_resources), + .resource = dm355_asp1_resources, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm355_io_desc[] = { @@ -732,6 +758,19 @@ static struct davinci_soc_info davinci_soc_info_dm355 = { .sram_len = SZ_32K, }; +void __init dm355_init_asp1(u32 evt_enable, struct snd_platform_data *pdata) +{ + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + if (evt_enable & ASP1_TX_EVT_EN) + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + + if (evt_enable & ASP1_RX_EVT_EN) + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + dm355_asp1_device.dev.platform_data = pdata; + platform_device_register(&dm355_asp1_device); +} + void __init dm355_init(void) { davinci_common_init(&davinci_soc_info_dm355); diff --git a/arch/arm/mach-davinci/dm644x.c b/arch/arm/mach-davinci/dm644x.c index 1b3aec8..351a131 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -27,6 +27,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -303,7 +304,7 @@ struct davinci_clk dm644x_clks[] = { CLK("davinci_emac.1", NULL, &emac_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK("palm_bk3710", NULL, &ide_clk), - CLK("soc-audio.0", NULL, &asp_clk), + CLK(NULL, "asp0", &asp_clk), CLK("davinci_mmc.0", NULL, &mmcsd_clk), CLK(NULL, "spi", &spi_clk), CLK(NULL, "gpio", &gpio_clk), @@ -549,6 +550,32 @@ static struct platform_device dm644x_edma_device = { .resource = edma_resources, }; +/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ +static struct resource dm644x_asp_resources[] = { + { + .start = DAVINCI_ASP0_BASE, + .end = DAVINCI_ASP0_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_ASP0_TX, + .end = DAVINCI_DMA_ASP0_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_ASP0_RX, + .end = DAVINCI_DMA_ASP0_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm644x_asp_device = { + .name = "davinci-asp", + .id = -1, + .num_resources = ARRAY_SIZE(dm644x_asp_resources), + .resource = dm644x_asp_resources, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm644x_io_desc[] = { @@ -666,6 +693,13 @@ static struct davinci_soc_info davinci_soc_info_dm644x = { .sram_len = SZ_16K, }; +void __init dm644x_init_asp(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM644X_MCBSP); + dm644x_asp_device.dev.platform_data = pdata; + platform_device_register(&dm644x_asp_device); +} + void __init dm644x_init(void) { davinci_common_init(&davinci_soc_info_dm644x); diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index e4d7d0f..591aecd 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -27,6 +27,7 @@ #include <mach/time.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -587,6 +588,66 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static struct resource dm646x_mcasp0_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + /* first TX, then RX */ + { + .start = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .end = DAVINCI_DM646X_DMA_MCASP0_AXEVT0, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .end = DAVINCI_DM646X_DMA_MCASP0_AREVT0, + .flags = IORESOURCE_DMA, + }, +}; + +static struct resource dm646x_mcasp1_resources[] = { + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + /* DIT mode, only TX event */ + { + .start = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .end = DAVINCI_DM646X_DMA_MCASP1_AXEVT1, + .flags = IORESOURCE_DMA, + }, + /* DIT mode, dummy entry */ + { + .start = -1, + .end = -1, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm646x_mcasp0_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(dm646x_mcasp0_resources), + .resource = dm646x_mcasp0_resources, +}; + +static struct platform_device dm646x_mcasp1_device = { + .name = "davinci-mcasp", + .id = 1, + .num_resources = ARRAY_SIZE(dm646x_mcasp1_resources), + .resource = dm646x_mcasp1_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "spdif-dit", + .id = -1, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm646x_io_desc[] = { @@ -697,6 +758,19 @@ static struct davinci_soc_info davinci_soc_info_dm646x = { .sram_len = SZ_32K, }; +void __init dm646x_init_mcasp0(struct snd_platform_data *pdata) +{ + dm646x_mcasp0_device.dev.platform_data = pdata; + platform_device_register(&dm646x_mcasp0_device); +} + +void __init dm646x_init_mcasp1(struct snd_platform_data *pdata) +{ + dm646x_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&dm646x_mcasp1_device); + platform_device_register(&dm646x_dit_device); +} + void __init dm646x_init(void) { davinci_common_init(&davinci_soc_info_dm646x); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..038ecb7 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -5,21 +5,52 @@ #define __ASM_ARCH_DAVINCI_ASP_H #include <mach/irqs.h> +#include <mach/edma.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 + /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT #define DAVINCI_ASP1_RX_INT IRQ_MBRINT #define DAVINCI_ASP1_TX_INT IRQ_MBXINT +struct snd_platform_data { + char *clk_name; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + #endif /* __ASM_ARCH_DAVINCI_ASP_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm355.h b/arch/arm/mach-davinci/include/mach/dm355.h index 54903b7..03faaef 100644 --- a/arch/arm/mach-davinci/include/mach/dm355.h +++ b/arch/arm/mach-davinci/include/mach/dm355.h @@ -12,11 +12,16 @@ #define __ASM_ARCH_DM355_H #include <mach/hardware.h> +#include <mach/asp.h> + +#define ASP1_TX_EVT_EN 1 +#define ASP1_RX_EVT_EN 2 struct spi_board_info; void __init dm355_init(void); void dm355_init_spi0(unsigned chipselect_mask, struct spi_board_info *info, unsigned len); +void __init dm355_init_asp1(u32 evt_enable, struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM355_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm644x.h b/arch/arm/mach-davinci/include/mach/dm644x.h index 15d42b9..8b157ce 100644 --- a/arch/arm/mach-davinci/include/mach/dm644x.h +++ b/arch/arm/mach-davinci/include/mach/dm644x.h @@ -25,6 +25,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM644X_EMAC_BASE (0x01C80000) #define DM644X_EMAC_CNTRL_OFFSET (0x0000) @@ -34,5 +35,6 @@ #define DM644X_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm644x_init(void); +void __init dm644x_init_asp(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM644X_H */ diff --git a/arch/arm/mach-davinci/include/mach/dm646x.h b/arch/arm/mach-davinci/include/mach/dm646x.h index 1fc764c..0585484 100644 --- a/arch/arm/mach-davinci/include/mach/dm646x.h +++ b/arch/arm/mach-davinci/include/mach/dm646x.h @@ -13,6 +13,7 @@ #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM646X_EMAC_BASE (0x01C80000) #define DM646X_EMAC_CNTRL_OFFSET (0x0000) @@ -22,5 +23,7 @@ #define DM646X_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm646x_init(void); +void __init dm646x_init_mcasp0(struct snd_platform_data *pdata); +void __init dm646x_init_mcasp1(struct snd_platform_data *pdata); #endif /* __ASM_ARCH_DM646X_H */ -- 1.5.6

3 2

[alsa-devel] [PATCH - quirk for USB Aureon cards 0/1] reworked naming
by Andrea Borgia 15 Jun '09

15 Jun '09

This patch changes yet again the ID for the TTA cards, resulting in a more reasonable name: 1 [Aureon51MkII ]: USB-Audio - Aureon5.1MkII TerraTec Aureon5.1MkII at usb-0000:00:03.0-2, full speed Andrea Borgia (1): rework quirk for TerraTec Aureon USB 5.1 MkII sound/usb/usbquirks.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

2 3

[alsa-devel] [PATCH 26/33] sound/ps3: Fix checkpatch issues
by Geert Uytterhoeven 15 Jun '09

15 Jun '09

Signed-off-by: Geert Uytterhoeven <Geert.Uytterhoeven(a)sonycom.com> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/ppc/snd_ps3.c | 32 +++++++++++++++----------------- 1 files changed, 15 insertions(+), 17 deletions(-) diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index f361c26..d660b0f 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -165,8 +165,7 @@ static const struct snd_pcm_hardware snd_ps3_pcm_hw = { .fifo_size = PS3_AUDIO_FIFO_SIZE }; -static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = -{ +static struct snd_pcm_ops snd_ps3_pcm_spdif_ops = { .open = snd_ps3_pcm_open, .close = snd_ps3_pcm_close, .prepare = snd_ps3_pcm_prepare, @@ -183,7 +182,7 @@ static int snd_ps3_verify_dma_stop(struct snd_ps3_card_info *card, int dma_ch, done, retries, stop_forced = 0; uint32_t status; - for (dma_ch = 0; dma_ch < 8; dma_ch ++) { + for (dma_ch = 0; dma_ch < 8; dma_ch++) { retries = count; do { status = read_reg(PS3_AUDIO_KICK(dma_ch)) & @@ -259,9 +258,7 @@ static void snd_ps3_kick_dma(struct snd_ps3_card_info *card) /* * convert virtual addr to ioif bus addr. */ -static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, - void * paddr, - int ch) +static dma_addr_t v_to_bus(struct snd_ps3_card_info *card, void *paddr, int ch) { return card->dma_start_bus_addr[ch] + (paddr - card->dma_start_vaddr[ch]); @@ -321,7 +318,7 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card, spin_lock_irqsave(&card->dma_lock, irqsave); for (ch = 0; ch < 2; ch++) { start_vaddr = card->dma_next_transfer_vaddr[0]; - for (stage = 0; stage < fill_stages; stage ++) { + for (stage = 0; stage < fill_stages; stage++) { dma_ch = stage * 2 + ch; if (silent) dma_addr = card->null_buffer_start_dma_addr; @@ -619,7 +616,7 @@ static int snd_ps3_change_avsetting(struct snd_ps3_card_info *card) PS3_AUDIO_AO_3WMCTRL_ASOEN(2) | PS3_AUDIO_AO_3WMCTRL_ASOEN(3)), 0); - wmb(); /* ensure the hardware sees the change */ + wmb(); /* ensure the hardware sees the change */ /* wait for actually stopped */ retries = 1000; while ((read_reg(PS3_AUDIO_AO_3WMCTRL) & @@ -798,20 +795,20 @@ static struct snd_kcontrol_new spdif_ctls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_cmask_get, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_pmask_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_PCM, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_ps3_spdif_mask_info, .get = snd_ps3_spdif_default_get, .put = snd_ps3_spdif_default_put, @@ -1020,11 +1017,12 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) * its size should be lager than PS3_AUDIO_FIFO_STAGE_SIZE * 2 * PAGE_SIZE is enogh */ - if (!(the_card.null_buffer_start_vaddr = - dma_alloc_coherent(&the_card.ps3_dev->core, - PAGE_SIZE, - &the_card.null_buffer_start_dma_addr, - GFP_KERNEL))) { + the_card.null_buffer_start_vaddr = + dma_alloc_coherent(&the_card.ps3_dev->core, + PAGE_SIZE, + &the_card.null_buffer_start_dma_addr, + GFP_KERNEL); + if (!the_card.null_buffer_start_vaddr) { pr_info("%s: nullbuffer alloc failed\n", __func__); goto clean_preallocate; } @@ -1148,7 +1146,7 @@ static irqreturn_t snd_ps3_interrupt(int irq, void *dev_id) SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL : SND_PS3_DMA_FILLTYPE_SILENT_RUNNING); snd_ps3_kick_dma(card); - card->silent --; + card->silent--; } else { snd_ps3_program_dma(card, (underflow_occured) ? -- 1.6.2.4

3 7

[alsa-devel] Crashing on x86_64
by WILMES Andreas 14 Jun '09

14 Jun '09

Hello guys, at first a big thank for your grateful work! My X-Fi card even sounds better than on my XP installation. I'm using Songbird on Kubuntu 8.04 with the latest snapshop. By increasing or fast pausing of songs my machine freezes. I can't even get any logs for that. Any ideas?

1 1

[alsa-devel] [PATCH, RFC 0/3] magician fix and UDA1380 i2c device registration change
by Philipp Zabel 14 Jun '09

14 Jun '09

Hi Mark, Patch 1/3 is a follow-up fix to commit a820532002e70e3a06f1ea7133e9b02443d07382 "ASoC: pxa-ssp.c fix clock/frame invert". Patch 2/3 removes the I2C board info registration from the UDA1380 driver. Instead, device instantiation will be done via global i2c_board_info from board files (patch 3/3). At the same time, platform specific configuration is moved to platform data. Common power/reset GPIO handling moves into the codec driver, which will allow to completely power down the codec depending on ASoC power state in the future.

1 0

[PATCH - quirk for USB Aureon cards 1/1] rework quirk for TerraTec Aureon USB 5.1 MkII
by Andrea Borgia 14 Jun '09

14 Jun '09

Signed-off-by: Andrea Borgia <andrea(a)borgia.bo.it> diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f0f7624..f6f201e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1989,7 +1989,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE(0x0ccd, 0x0028), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "TerraTec", - .product_name = "Aureon 5.1 MkII", + .product_name = "Aureon5.1MkII", .ifnum = QUIRK_NO_INTERFACE } }, -- 1.6.0.4 --------------000205060801080305030206--

1 0

[alsa-devel] Maybe the build script for Alsa Libs does not properly install for 64-bit systems? (alsa-lib-1.0.20.tar.bz2)
by bill＠tellytopia.com 14 Jun '09

14 Jun '09

Maybe the build script for Alsa Libs does not properly install for 64-bit systems? (alsa-lib-1.0.20.tar.bz2) Built alsa-lib and did make install; built alsa-driver and did make install; tried to build alsa-utils and found Compile errors when building on a 64-bit platform in Fedora Core 9. Notice the old date on /lib64/libasound.so.2.0.0: [root@localhost alsa]# ls -l /lib64/libasound* lrwxrwxrwx 1 root root 18 2009-06-12 14:29 /lib64/libasound.so.2 -> libasound.so.2.0.0 -rwxr-xr-x 1 root root 920640 2008-04-04 01:26 /lib64/libasound.so.2.0.0 [root@localhost alsa]# ls -l /usr/lib/libasound* -rwxr-xr-x 1 root root 813 2009-06-12 17:02 /usr/lib/libasound.la lrwxrwxrwx 1 root root 18 2009-06-12 17:02 /usr/lib/libasound.so -> libasound.so.2.0.0 lrwxrwxrwx 1 root root 18 2009-06-12 17:02 /usr/lib/libasound.so.2 -> libasound.so.2.0.0 -rwxr-xr-x 1 root root 3544926 2009-06-12 17:02 /usr/lib/libasound.so.2.0.0 Here is the system this is built on: [root@localhost alsa-utils-1.0.20]# uname -a Linux localhost.localdomain 2.6.25-14.fc9.x86_64 #1 SMP Thu May 1 06:06:21 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Here is the error when compiling utils: --------------------------------------- make[1]: Entering directory `/root/Desktop/alsa/alsa-utils-1.0.20/aplay' if gcc -DHAVE_CONFIG_H -I. -I. -I../include -I../include -g -O2 -MT aplay.o -MD -MP -MF ".deps/aplay.Tpo" -c -o aplay.o aplay.c; \ then mv -f ".deps/aplay.Tpo" ".deps/aplay.Po"; else rm -f ".deps/aplay.Tpo"; exit 1; fi gcc -g -O2 -o aplay aplay.o -lrt -lasound -lm -ldl -lpthread aplay.o: In function `set_params': /root/Desktop/alsa/alsa-utils-1.0.20/aplay/aplay.c:1038: undefined reference to `snd_pcm_hw_params_is_monotonic' aplay.o: In function `do_test_position': /root/Desktop/alsa/alsa-utils-1.0.20/aplay/aplay.c:1455: undefined reference to `snd_pcm_avail_delay' collect2: ld returned 1 exit status make[1]: *** [aplay] Error 1 make[1]: Leaving directory `/root/Desktop/alsa/alsa-utils-1.0.20/aplay' make: *** [all-recursive] Error 1 [root@localhost alsa-utils-1.0.20]# So then link the /lib64 libraries to the /usr/lib equivalents and cross fingers: [root@localhost lib64]# ls -l libasound* lrwxrwxrwx 1 root root 27 2009-06-13 01:25 libasound.2.0.0 -> /usr/lib/libasound.so.2.0.0 lrwxrwxrwx 1 root root 27 2009-06-13 01:26 libasound.so.2 -> /usr/lib/libasound.so.2.0.0 -rwxr-xr-x 1 root root 923208 2008-04-04 01:26 libasound.so.2.0.0-original Then compile utils again. No compile errors. So the compile gave no errors when the "old" /lib64/libasound.so.2 was not used, and the /usr/lib/libasound.so.2 was used instead. Does this mean that the build script for alsa-lib-1.0.20 did not put in the correct library? Or what? Thanks for thoughtful responses. Bill

2 1

[alsa-devel] Request for Device ID addition : 6935G
by Karthik Ramgopal 14 Jun '09

14 Jun '09

Hi, Acer Aspire 6935G sound system is confirmed to be functional with model=acer-aspire-8930g. Kindly add device ID 0x1025 0x0146 to the quirks list in patch_realtek.c . Jack detetction and playback is working without any hda-verbing. Just the internal mic seems to be a problem. Can't hear my voice, can just hear static when I playback a recording. Anyways big thanks to Hector and Takashi for getting it all to work. Regards, Karthik

3 3

Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Jean-Pierre André 14 Jun '09

14 Jun '09

Hi, > At Fri, 12 Jun 2009 15:01:09 -0300, > Gustavo Maciel Dias Vieira wrote: > >> > >> > Em Qui, 2009-06-11 às 21:27 +0200, Takashi Iwai escreveu: >> >>>> > > > Interrupt 22 stays always with 0. Even when correctly playing >>>> >>> > > sounds. I >>> >>>> > > > tested this on another machine with HD-Audio, and the equivalent >>>> > > > interrupt increased when playing. Also, in this other system >>>> >>> > > interrupt 0 >>> >>>> > > > (timer) is hardly issued, but in this laptop it is firing all the >>>> >>> > > time. >>> >>>> > > > >>>> > > > How can the driver be working without interrupts? >>>> >>> > > >>> > > That's weird. >>> > > >>> > > Did you try enable_msi=1 module option, BTW? >>> >> > >> > Just tried now. The interrupts started flowing and, tada, jack sense >> > works! >> > >> > Thanks, Takashi, that solves it for me. Is there a way to automate this >> > in the driver for the other less technically inclined users? >> > > Well, as I mentioned, it's basically a bug in the lower layer like > ACPI or BIOS. So, the best is to fix them. Using MSI is a kind of > workaround, and I'm wondering whether this is specific to your device > or in general with other HP dv4, since I didn't hear of the similar > bug reports. > > > Takashi > Weird indeed. Actually I had already reported this behaviour, see http://article.gmane.org/gmane.linux.alsa.devel/59809 So far I have not been able to get the interrupts at IRQ22. They still occur at IRQ20. However the problem is hidden when using boot options irqpoll or noapic. This is not a hardware or bios problem, as the sound is correct on Windows. Initially I was using kernel 2.6.27 and alsa-driver 1.0.19, currently using kernel 2.6.29 and alsa-driver 1.0.20 with the same behaviour. I have tried numerous modprobe options. Currently I am using : alias snd-card-0 snd-hda-intel options sound slots=snd-hda-intel options snd-hda-intel model=hp-m4 enable_msi=1 bdl_pos_adj=32 probe_mask=0x1ff As Gustavo seems to have succeeded, can he post kernel and alsa-driver versions, boot parameters and modprobe parameters ? Thanks in advance Jean-Pierre

2 1

[alsa-devel] [RFC PATCH] ASoC: au1x: platform driver restructuring
by Manuel Lauss 13 Jun '09

13 Jun '09

Hi Mark! Please have a look at the patch below and verify it does what you suggested earlier, and if you have no objections, add it to your patchqueue. Both the AC97 and I2S modules are converted to full platform drivers; the PCM (DMA) interface is initialized during platform probe. Run-tested on DB1200 board. Thank you! Manuel Lauss --- From: Manuel Lauss <manuel.lauss(a)gmail.com> Subject: [PATCH] ASoC: au1x: platform driver restructuring Make the AC97/I2S modules full platform drivers and update the sample machine code. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- sound/soc/au1x/dbdma2.c | 24 +++++----- sound/soc/au1x/psc-ac97.c | 103 +++++++++++++++++++++++++----------------- sound/soc/au1x/psc-i2s.c | 103 ++++++++++++++++++++++++------------------ sound/soc/au1x/psc.h | 7 ++- sound/soc/au1x/sample-ac97.c | 48 +++++++++++-------- 5 files changed, 165 insertions(+), 120 deletions(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 594c6c5..10d21ec 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -331,7 +331,7 @@ static int au1xpsc_pcm_new(struct snd_card *card, return 0; } -static int au1xpsc_pcm_probe(struct platform_device *pdev) +int au1xpsc_asoc_dbdma_init(struct platform_device *pdev) { struct resource *r; int ret; @@ -375,8 +375,9 @@ out1: au1xpsc_audio_pcmdma[PCM_TX] = NULL; return ret; } +EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_init); -static int au1xpsc_pcm_remove(struct platform_device *pdev) +void au1xpsc_asoc_dbdma_exit(void) { int i; @@ -387,36 +388,33 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev) au1xpsc_audio_pcmdma[i] = NULL; } } - - return 0; } +EXPORT_SYMBOL_GPL(au1xpsc_asoc_dbdma_exit); /* au1xpsc audio platform */ struct snd_soc_platform au1xpsc_soc_platform = { .name = "au1xpsc-pcm-dbdma", - .probe = au1xpsc_pcm_probe, - .remove = au1xpsc_pcm_remove, .pcm_ops = &au1xpsc_pcm_ops, .pcm_new = au1xpsc_pcm_new, .pcm_free = au1xpsc_pcm_free_dma_buffers, }; EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); -static int __init au1xpsc_audio_dbdma_init(void) +static int __init au1xpsc_audio_dbdma_load(void) { au1xpsc_audio_pcmdma[PCM_TX] = NULL; au1xpsc_audio_pcmdma[PCM_RX] = NULL; return snd_soc_register_platform(&au1xpsc_soc_platform); } -static void __exit au1xpsc_audio_dbdma_exit(void) +static void __exit au1xpsc_audio_dbdma_unload(void) { snd_soc_unregister_platform(&au1xpsc_soc_platform); } -module_init(au1xpsc_audio_dbdma_init); -module_exit(au1xpsc_audio_dbdma_exit); +module_init(au1xpsc_audio_dbdma_load); +module_exit(au1xpsc_audio_dbdma_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..ee8e642 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -236,8 +236,32 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, return ret; } -static int au1xpsc_ac97_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .ac97_control = 1, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xpsc_ac97_dai_ops, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + + +static int __devinit au1xpsc_ac97_probe(struct platform_device *pdev) { int ret; struct resource *r; @@ -287,8 +311,18 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, * there may not be any codec clock yet. */ - return 0; + ret = au1xpsc_asoc_dbdma_init(pdev); + if (ret) + goto out2; + + ret = snd_soc_register_dai(&au1xpsc_ac97_dai); + if (!ret) + return 0; + + au1xpsc_asoc_dbdma_exit(); +out2: + iounmap(au1xpsc_ac97_workdata->mmio); out1: release_resource(au1xpsc_ac97_workdata->ioarea); kfree(au1xpsc_ac97_workdata->ioarea); @@ -298,9 +332,11 @@ out0: return ret; } -static void au1xpsc_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_ac97_remove(struct platform_device *pdev) { + snd_soc_unregister_dai(&au1xpsc_ac97_dai); + au1xpsc_asoc_dbdma_exit(); + /* disable PSC completely */ au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); au_sync(); @@ -312,9 +348,11 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev, kfree(au1xpsc_ac97_workdata->ioarea); kfree(au1xpsc_ac97_workdata); au1xpsc_ac97_workdata = NULL; + + return 0; } -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) +static int au1xpsc_ac97_suspend(struct platform_device *pdev, pm_message_t m) { /* save interesting registers and disable PSC */ au1xpsc_ac97_workdata->pm[0] = @@ -328,7 +366,7 @@ static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) return 0; } -static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) +static int au1xpsc_ac97_resume(struct platform_device *pdev) { /* restore PSC clock config */ au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, @@ -342,48 +380,31 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, -}; - -struct snd_soc_dai au1xpsc_ac97_dai = { - .name = "au1xpsc_ac97", - .ac97_control = 1, - .probe = au1xpsc_ac97_probe, - .remove = au1xpsc_ac97_remove, - .suspend = au1xpsc_ac97_suspend, - .resume = au1xpsc_ac97_resume, - .playback = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .owner = THIS_MODULE, }, - .capture = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, - }, - .ops = &au1xpsc_ac97_dai_ops, + .probe = au1xpsc_ac97_probe, + .remove = __devexit_p(au1xpsc_ac97_remove), + .suspend = au1xpsc_ac97_suspend, + .resume = au1xpsc_ac97_resume, }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); -static int __init au1xpsc_ac97_init(void) +static int __init au1xpsc_ac97_load(void) { au1xpsc_ac97_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_ac97_dai); + return platform_driver_register(&au1xpsc_ac97_driver); } -static void __exit au1xpsc_ac97_exit(void) +static void __exit au1xpsc_ac97_unload(void) { - snd_soc_unregister_dai(&au1xpsc_ac97_dai); + platform_driver_unregister(&au1xpsc_ac97_driver); } -module_init(au1xpsc_ac97_init); -module_exit(au1xpsc_ac97_exit); +module_init(au1xpsc_ac97_load); +module_exit(au1xpsc_ac97_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index bb58932..f830f5a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -262,8 +262,31 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int au1xpsc_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = &au1xpsc_i2s_dai_ops, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int au1xpsc_i2s_probe(struct platform_device *pdev) { struct resource *r; unsigned long sel; @@ -294,9 +317,7 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, if (!au1xpsc_i2s_workdata->mmio) goto out1; - /* preserve PSC clock source set up by platform (dev.platform_data - * is already occupied by soc layer) - */ + /* preserve PSC clock source set up by platform */ sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); au_sync(); @@ -312,9 +333,16 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, * be running yet; depending on clock input for PSC a wait might * time out. */ + ret = au1xpsc_asoc_dbdma_init(pdev); + if (ret) + goto out2; - return 0; + ret = snd_soc_register_dai(&au1xpsc_i2s_dai); + if (!ret) + return 0; +out2: + iounmap(au1xpsc_i2s_workdata->mmio); out1: release_resource(au1xpsc_i2s_workdata->ioarea); kfree(au1xpsc_i2s_workdata->ioarea); @@ -324,9 +352,11 @@ out0: return ret; } -static void au1xpsc_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_i2s_remove(struct platform_device *pdev) { + snd_soc_unregister_dai(&au1xpsc_i2s_dai); + au1xpsc_asoc_dbdma_exit(); + au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); au_sync(); au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); @@ -337,9 +367,11 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev, kfree(au1xpsc_i2s_workdata->ioarea); kfree(au1xpsc_i2s_workdata); au1xpsc_i2s_workdata = NULL; + + return 0; } -static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_suspend(struct platform_device *pdev, pm_message_t m) { /* save interesting register and disable PSC */ au1xpsc_i2s_workdata->pm[0] = @@ -353,7 +385,7 @@ static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) return 0; } -static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_resume(struct platform_device *pdev) { /* select I2S mode and PSC clock */ au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); @@ -367,48 +399,31 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, -}; - -struct snd_soc_dai au1xpsc_i2s_dai = { - .name = "au1xpsc_i2s", - .probe = au1xpsc_i2s_probe, - .remove = au1xpsc_i2s_remove, - .suspend = au1xpsc_i2s_suspend, - .resume = au1xpsc_i2s_resume, - .playback = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .owner = THIS_MODULE, }, - .capture = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .ops = &au1xpsc_i2s_dai_ops, + .probe = au1xpsc_i2s_probe, + .remove = __devexit_p(au1xpsc_i2s_remove), + .suspend = au1xpsc_i2s_suspend, + .resume = au1xpsc_i2s_resume, }; -EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_init(void) +static int __init au1xpsc_i2s_load(void) { au1xpsc_i2s_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_i2s_dai); + return platform_driver_register(&au1xpsc_i2s_driver); } -static void __exit au1xpsc_i2s_exit(void) +static void __exit au1xpsc_i2s_unload(void) { - snd_soc_unregister_dai(&au1xpsc_i2s_dai); + platform_driver_unregister(&au1xpsc_i2s_driver); } -module_init(au1xpsc_i2s_init); -module_exit(au1xpsc_i2s_exit); +module_init(au1xpsc_i2s_load); +module_exit(au1xpsc_i2s_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a0..9ee4f59 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -21,6 +21,9 @@ extern struct snd_soc_dai au1xpsc_i2s_dai; extern struct snd_soc_platform au1xpsc_soc_platform; extern struct snd_ac97_bus_ops soc_ac97_ops; +extern int au1xpsc_asoc_dbdma_init(struct platform_device *pdev); +extern void au1xpsc_asoc_dbdma_exit(void); + struct au1xpsc_audio_data { void __iomem *mmio; diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c index 27683eb..86380f6 100644 --- a/sound/soc/au1x/sample-ac97.c +++ b/sound/soc/au1x/sample-ac97.c @@ -1,7 +1,7 @@ /* * Sample Au12x0/Au1550 PSC AC97 sound machine. * - * Copyright (c) 2007-2008 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * Copyright (c) 2007-2009 Manuel Lauss <manuel.lauss(a)gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms outlined in the file COPYING at the root of this @@ -44,13 +44,13 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { static struct snd_soc_card au1xpsc_sample_ac97_machine = { .name = "Au1xxx PSC AC97 Audio", + .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .dai_link = &au1xpsc_sample_ac97_dai, .num_links = 1, }; static struct snd_soc_device au1xpsc_sample_ac97_devdata = { .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .codec_dev = &soc_codec_dev_ac97, }; @@ -82,7 +82,14 @@ static struct resource au1xpsc_psc1_res[] = { }, }; -static struct platform_device *au1xpsc_sample_ac97_dev; +static struct platform_device psc_ac97_dev = { + .name = "au1xpsc_ac97", + .id = 0, + .resource = au1xpsc_psc1_res, + .num_resources = ARRAY_SIZE(au1xpsc_psc1_res), +}; + +static struct platform_device *au1xpsc_sample_asoc_dev; static int __init au1xpsc_sample_ac97_load(void) { @@ -99,41 +106,42 @@ static int __init au1xpsc_sample_ac97_load(void) au_sync(); #endif - ret = -ENOMEM; - /* setup PSC clock source for AC97 part: external clock provided * by codec. The psc-ac97.c driver depends on this setting! */ au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); au_sync(); - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; + /* register the PSC-AC97 platform device */ + ret = platform_device_register(&psc_ac97_dev); + if (ret) + return -ENODEV; - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; + /* register machine fabric with ASoC core */ + ret = -ENOMEM; + au1xpsc_sample_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!au1xpsc_sample_asoc_dev) + goto out; - platform_set_drvdata(au1xpsc_sample_ac97_dev, + platform_set_drvdata(au1xpsc_sample_asoc_dev, &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); + au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_asoc_dev->dev; + ret = platform_device_add(au1xpsc_sample_asoc_dev); if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; + platform_device_put(au1xpsc_sample_asoc_dev); + au1xpsc_sample_asoc_dev = NULL; } out: + platform_device_unregister(&psc_ac97_dev); return ret; } static void __exit au1xpsc_sample_ac97_exit(void) { - platform_device_unregister(au1xpsc_sample_ac97_dev); + platform_device_unregister(au1xpsc_sample_asoc_dev); + platform_device_unregister(&psc_ac97_dev); } module_init(au1xpsc_sample_ac97_load); @@ -141,4 +149,4 @@ module_exit(au1xpsc_sample_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); -- 1.6.3.1

2 3

[PATCH] ASoC: Add stub suspend and resume calls for ASoC subdevices
by Mark Brown 13 Jun '09

13 Jun '09

Now that ASoC subdevices can be regular devices they can have normal suspend and resume calls from their buses. However, suspending them individually is not desirable since this can lead to problems such as pops and clicks from devices being suspended with their signals being amplified or clocks being stopped suddenly. This will be resolved by having the normal device model suspend and resume calls call into ASoC which will suspend the entire card while any of its components are suspended. At present this is not yet implemented but in order to aid the transition of drivers to the standard device model this patch adds API calls for the notifications. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 39 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5297ba7..e6704c0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -192,6 +192,11 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +#ifdef CONFIG_PM +int snd_soc_suspend_device(struct device *dev); +int snd_soc_resume_device(struct device *dev); +#endif + /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..4414117 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -788,6 +788,45 @@ static int soc_resume(struct platform_device *pdev) return 0; } +/** + * snd_soc_suspend_device: Notify core of device suspend + * + * @dev: Device being suspended. + * + * In order to ensure that the entire audio subsystem is suspended in a + * coordinated fashion ASoC devices should suspend themselves when + * called by ASoC. When the standard kernel suspend process asks the + * device to suspend it should call this function to initiate a suspend + * of the entire ASoC card. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_suspend_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_suspend_device); + +/** + * snd_soc_resume_device: Notify core of device resume + * + * @dev: Device being resumed. + * + * In order to ensure that the entire audio subsystem is resumed in a + * coordinated fashion ASoC devices should resume themselves when called + * by ASoC. When the standard kernel resume process asks the device + * to resume it should call this function. Once all the components of + * the card have notified that they are ready to be resumed the card + * will be resumed. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_resume_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_resume_device); + #else #define soc_suspend NULL #define soc_resume NULL -- 1.6.3.1

1 0