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

Re: [alsa-devel] Recent breakages (was: (no subject))
by James Gardiner 31 May '09

31 May '09

--- On Sun, 5/31/09, Takashi Iwai <tiwai(a)suse.de> wrote: > OK, I found out the culprit. It was a typo in my last > patch. > Fixed now. > > Sorry for inconvenience. > > > Takashi > Working now! Thank you! James Gardiner

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[alsa-devel] Recent breakages (was: (no subject))
by renidragsemaj＠yahoo.com 31 May '09

31 May '09

--- On Sat, 5/30/09, Takashi Iwai <tiwai(a)suse.de> wrote: >> Hi, someone recently submitted changes to the patch_sigmatel.c codec >> source. Whatever was done is breaking my patch for the HP dv4-1222nr >> laptop. I also strongly suspect that other code in the stac92hd71bxx >> function has been affected. There is no autodetect, there is no >> option for modprobe that will load that codec currently, to the best >> of my knowledge. I've tracked the difference to snapshot >> 20090529. Before that, I can get it to work, there and after refuses >> to work properly. > >Could you try to copy hda_intel.c from 1.0.20 and check whether this >fixes the problem? Most likely I broke something. You seem to have that right. I changed out that file like you said and it went back to working. Now, only to find out what happened... >> >> Also, there was a recent change to the >> stac92xx_hp_check_power_status function: While the change looks good >> for its intended purpose of localizing board changes to the >> switch/case statements in the stac92hd71bxx function, the >> implementor reversed the if/then selector: > >Fixed now. Thanks for checking! Thank you. It seems that I had two big booboos with this part of my post. 1) Sorry about all those question marks: It looks like my cut and paste job didn't go the way I planned. 2) I messed up and switched the old and new code in my post, but it looks like you caught my screw-up. The recent snapshot (20090531) has it right. *Don't change it back!* :) Thank you, James Gardiner

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[alsa-devel] [PATCH 1/5] ALSA: snd_usb_caiaq: clean whitespaces
by Daniel Mack 30 May '09

30 May '09

Cosmetic changes only, no code change. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 88 +++++++++++++++++++++++----------------------- sound/usb/caiaq/device.c | 60 ++++++++++++++++---------------- sound/usb/caiaq/midi.c | 24 ++++++------ 3 files changed, 86 insertions(+), 86 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b13ce76..b144513 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -42,10 +42,10 @@ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S24_3BE, - .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 0, /* will overwrite later */ @@ -68,7 +68,7 @@ activate_substream(struct snd_usb_caiaqdev *dev, dev->sub_capture[sub->number] = sub; } -static void +static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { @@ -118,7 +118,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return -EPIPE; } } - + return 0; } @@ -129,7 +129,7 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; - + dev->streaming = 0; for (i = 0; i < N_URBS; i++) { @@ -154,7 +154,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { - /* when the last client has stopped streaming, + /* when the last client has stopped streaming, * all sample rates are allowed again */ stream_stop(dev); dev->pcm_info.rates = dev->samplerates; @@ -194,7 +194,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; debug("%s(%p)\n", __func__, substream); - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -205,19 +205,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (dev->streaming) return 0; - + /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) dev->pcm_info.rates = 1 << i; - + snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; if (dev->spec.data_alignment == 2) bytes_per_sample++; - + bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; @@ -232,7 +232,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) ret = stream_start(dev); if (ret) return ret; - + dev->output_running = 0; wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); if (!dev->output_running) { @@ -273,7 +273,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) return SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + return bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else return bytes_to_frames(sub->runtime, @@ -291,7 +291,7 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer }; - + static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream **subs) { @@ -333,7 +333,7 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) @@ -354,14 +354,14 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { - for (stream = 0; - stream < dev->n_streams; + for (stream = 0; + stream < dev->n_streams; stream++, i++) { if (dev->first_packet) continue; check_byte = MAKE_CHECKBYTE(dev, stream, i); - + if ((usb_buf[i] & 0x3f) != check_byte) dev->input_panic = 1; @@ -410,21 +410,21 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", + debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); dev->warned = 1; } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, +static void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - + for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; @@ -444,7 +444,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == + i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); @@ -453,7 +453,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { - struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; @@ -478,7 +478,7 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; - + if (len > 0) { spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); @@ -497,14 +497,14 @@ static void read_completed(struct urb *urb) out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } - + /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; urb->iso_frame_desc[frame].length = BYTES_PER_FRAME; urb->iso_frame_desc[frame].actual_length = 0; } - + urb->number_of_packets = FRAMES_PER_URB; urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); @@ -528,7 +528,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) struct usb_device *usb_dev = dev->chip.dev; unsigned int pipe; - pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); @@ -547,25 +547,25 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) return urbs; } - urbs[i]->transfer_buffer = + urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { log("unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } - + for (frame = 0; frame < FRAMES_PER_URB; frame++) { - struct usb_iso_packet_descriptor *iso = + struct usb_iso_packet_descriptor *iso = &urbs[i]->iso_frame_desc[frame]; - + iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - + urbs[i]->dev = usb_dev; urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB + urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; @@ -589,7 +589,7 @@ static void free_urbs(struct urb **urbs) for (i = 0; i < N_URBS; i++) { if (!urbs[i]) continue; - + usb_kill_urb(urbs[i]); kfree(urbs[i]->transfer_buffer); usb_free_urb(urbs[i]); @@ -602,11 +602,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + dev->n_audio_in = max(dev->spec.num_analog_audio_in, + dev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + dev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); @@ -619,7 +619,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, + ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, dev->n_audio_out, dev->n_audio_in, &dev->pcm); if (ret < 0) { @@ -632,7 +632,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - + memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); @@ -651,9 +651,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); snd_pcm_lib_preallocate_pages_for_all(dev->pcm, @@ -662,7 +662,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); dev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); if (!dev->data_cb_info) @@ -672,14 +672,14 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; } - + dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); free_urbs(dev->data_urbs_in); return ret; } - + dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 515de1c..9be0f2e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -79,7 +79,7 @@ static struct usb_device_id snd_usb_id_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, - .idProduct = USB_PID_RIGKONTROL2 + .idProduct = USB_PID_RIGKONTROL2 }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, @@ -197,7 +197,7 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, if (buffer && len > 0) memcpy(dev->ep1_out_buf+1, buffer, len); - + dev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), dev->ep1_out_buf, len+1, &actual_len, 200); @@ -208,7 +208,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, { int ret; char tmp[5]; - + switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; case 48000: tmp[0] = SAMPLERATE_48000; break; @@ -237,12 +237,12 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (ret) return ret; - - if (!wait_event_timeout(dev->ep1_wait_queue, + + if (!wait_event_timeout(dev->ep1_wait_queue, dev->audio_parm_answer >= 0, HZ)) return -EPIPE; - - if (dev->audio_parm_answer != 1) + + if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); else dev->bpp = bpp; @@ -250,8 +250,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, - int digital, int analog, int erp) +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, + int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, @@ -262,7 +262,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) { int ret; char val[4]; - + /* device-specific startup specials */ switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): @@ -314,7 +314,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) dev->control_state, 1); break; } - + if (dev->spec.num_analog_audio_out + dev->spec.num_analog_audio_in + dev->spec.num_digital_audio_out + @@ -323,7 +323,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) if (ret < 0) log("Unable to set up audio system (ret=%d)\n", ret); } - + if (dev->spec.num_midi_in + dev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(dev); @@ -363,7 +363,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) if (devnum >= SNDRV_CARDS) return -ENODEV; - err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) return err; @@ -386,7 +386,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; int err, len; - + if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); return -EIO; @@ -395,19 +395,19 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_init_urb(&dev->ep1_in_urb); usb_init_urb(&dev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, + dev->ep1_in_buf, EP1_BUFSIZE, usb_ep1_command_reply_dispatch, dev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, + dev->midi_out_buf, EP1_BUFSIZE, snd_usb_caiaq_midi_output_done, dev); - + init_waitqueue_head(&dev->ep1_wait_queue); init_waitqueue_head(&dev->prepare_wait_queue); - + if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; @@ -420,10 +420,10 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iManufacturer, dev->vendor_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, dev->serial, CAIAQ_USB_STR_LEN); @@ -431,7 +431,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) c = strchr(dev->serial, ' '); if (c) *c = '\0'; - + strcpy(card->driver, MODNAME); strcpy(card->shortname, dev->product_name); @@ -449,18 +449,18 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) return 0; } -static int __devinit snd_probe(struct usb_interface *intf, +static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - + ret = create_card(device, &card); - + if (ret < 0) return ret; - + usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { @@ -468,7 +468,7 @@ static int __devinit snd_probe(struct usb_interface *intf, snd_card_free(card); return ret; } - + return 0; } @@ -489,10 +489,10 @@ static void snd_disconnect(struct usb_interface *intf) snd_usb_caiaq_input_free(dev); #endif snd_usb_caiaq_audio_free(dev); - + usb_kill_urb(&dev->ep1_in_urb); usb_kill_urb(&dev->midi_out_urb); - + snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); } diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 8fa8cd8..538e8c0 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -40,7 +40,7 @@ static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *subst if (!dev) return; - + dev->midi_receive_substream = up ? substream : NULL; } @@ -64,18 +64,18 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, struct snd_rawmidi_substream *substream) { int len, ret; - + dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, EP1_BUFSIZE - 3); - + if (len <= 0) return; - + dev->midi_out_buf[2] = len; dev->midi_out_urb.transfer_buffer_length = len+3; - + ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," @@ -88,7 +88,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - + if (up) { dev->midi_out_substream = substream; if (!dev->midi_out_active) @@ -113,12 +113,12 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len) { if (!dev->midi_receive_substream) return; - + snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } @@ -142,16 +142,16 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (device->spec.num_midi_out > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usb_caiaq_midi_output); } if (device->spec.num_midi_in > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usb_caiaq_midi_input); } - + device->rmidi = rmidi; return 0; @@ -160,7 +160,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - + dev->midi_out_active = 0; if (urb->status != 0) return; -- 1.6.3.1

2 6

[alsa-devel] [PATCH 1/5] ALSA: snd_usb_caiaq: clean whitespaces
by Daniel Mack 30 May '09

30 May '09

Cosmetic changes only, no code change. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 88 +++++++++++++++++++++++----------------------- sound/usb/caiaq/device.c | 60 ++++++++++++++++---------------- sound/usb/caiaq/midi.c | 24 ++++++------ 3 files changed, 86 insertions(+), 86 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b13ce76..b144513 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -42,10 +42,10 @@ (stream << 1) | (~(i / (dev->n_streams * BYTES_PER_SAMPLE_USB)) & 1) static struct snd_pcm_hardware snd_usb_caiaq_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S24_3BE, - .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000), .rate_min = 44100, .rate_max = 0, /* will overwrite later */ @@ -68,7 +68,7 @@ activate_substream(struct snd_usb_caiaqdev *dev, dev->sub_capture[sub->number] = sub; } -static void +static void deactivate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { @@ -118,7 +118,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) return -EPIPE; } } - + return 0; } @@ -129,7 +129,7 @@ static void stream_stop(struct snd_usb_caiaqdev *dev) debug("%s(%p)\n", __func__, dev); if (!dev->streaming) return; - + dev->streaming = 0; for (i = 0; i < N_URBS; i++) { @@ -154,7 +154,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (all_substreams_zero(dev->sub_playback) && all_substreams_zero(dev->sub_capture)) { - /* when the last client has stopped streaming, + /* when the last client has stopped streaming, * all sample rates are allowed again */ stream_stop(dev); dev->pcm_info.rates = dev->samplerates; @@ -194,7 +194,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; debug("%s(%p)\n", __func__, substream); - + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; @@ -205,19 +205,19 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) if (dev->streaming) return 0; - + /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ for (i=0; i < ARRAY_SIZE(rates); i++) if (runtime->rate == rates[i]) dev->pcm_info.rates = 1 << i; - + snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; if (dev->spec.data_alignment == 2) bytes_per_sample++; - + bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) * bytes_per_sample * CHANNELS_PER_STREAM * dev->n_streams; @@ -232,7 +232,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) ret = stream_start(dev); if (ret) return ret; - + dev->output_running = 0; wait_event_timeout(dev->prepare_wait_queue, dev->output_running, HZ); if (!dev->output_running) { @@ -273,7 +273,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) return SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + return bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else return bytes_to_frames(sub->runtime, @@ -291,7 +291,7 @@ static struct snd_pcm_ops snd_usb_caiaq_ops = { .trigger = snd_usb_caiaq_pcm_trigger, .pointer = snd_usb_caiaq_pcm_pointer }; - + static void check_for_elapsed_periods(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream **subs) { @@ -333,7 +333,7 @@ static void read_in_urb_mode0(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) @@ -354,14 +354,14 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (i = 0; i < iso->actual_length;) { if (i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == 0) { - for (stream = 0; - stream < dev->n_streams; + for (stream = 0; + stream < dev->n_streams; stream++, i++) { if (dev->first_packet) continue; check_byte = MAKE_CHECKBYTE(dev, stream, i); - + if ((usb_buf[i] & 0x3f) != check_byte) dev->input_panic = 1; @@ -410,21 +410,21 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } if ((dev->input_panic || dev->output_panic) && !dev->warned) { - debug("streaming error detected %s %s\n", + debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); dev->warned = 1; } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, +static void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; int stream, i; - + for (i = 0; i < iso->length;) { for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; @@ -444,7 +444,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && - i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == + i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); @@ -453,7 +453,7 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, static void read_completed(struct urb *urb) { - struct snd_usb_caiaq_cb_info *info = urb->context; + struct snd_usb_caiaq_cb_info *info = urb->context; struct snd_usb_caiaqdev *dev; struct urb *out; int frame, len, send_it = 0, outframe = 0; @@ -478,7 +478,7 @@ static void read_completed(struct urb *urb) out->iso_frame_desc[outframe].length = len; out->iso_frame_desc[outframe].actual_length = 0; out->iso_frame_desc[outframe].offset = BYTES_PER_FRAME * frame; - + if (len > 0) { spin_lock(&dev->spinlock); fill_out_urb(dev, out, &out->iso_frame_desc[outframe]); @@ -497,14 +497,14 @@ static void read_completed(struct urb *urb) out->transfer_flags = URB_ISO_ASAP; usb_submit_urb(out, GFP_ATOMIC); } - + /* re-submit inbound urb */ for (frame = 0; frame < FRAMES_PER_URB; frame++) { urb->iso_frame_desc[frame].offset = BYTES_PER_FRAME * frame; urb->iso_frame_desc[frame].length = BYTES_PER_FRAME; urb->iso_frame_desc[frame].actual_length = 0; } - + urb->number_of_packets = FRAMES_PER_URB; urb->transfer_flags = URB_ISO_ASAP; usb_submit_urb(urb, GFP_ATOMIC); @@ -528,7 +528,7 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) struct usb_device *usb_dev = dev->chip.dev; unsigned int pipe; - pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? + pipe = (dir == SNDRV_PCM_STREAM_PLAYBACK) ? usb_sndisocpipe(usb_dev, ENDPOINT_PLAYBACK) : usb_rcvisocpipe(usb_dev, ENDPOINT_CAPTURE); @@ -547,25 +547,25 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) return urbs; } - urbs[i]->transfer_buffer = + urbs[i]->transfer_buffer = kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); if (!urbs[i]->transfer_buffer) { log("unable to kmalloc() transfer buffer, OOM!?\n"); *ret = -ENOMEM; return urbs; } - + for (frame = 0; frame < FRAMES_PER_URB; frame++) { - struct usb_iso_packet_descriptor *iso = + struct usb_iso_packet_descriptor *iso = &urbs[i]->iso_frame_desc[frame]; - + iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - + urbs[i]->dev = usb_dev; urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB + urbs[i]->transfer_buffer_length = FRAMES_PER_URB * BYTES_PER_FRAME; urbs[i]->context = &dev->data_cb_info[i]; urbs[i]->interval = 1; @@ -589,7 +589,7 @@ static void free_urbs(struct urb **urbs) for (i = 0; i < N_URBS; i++) { if (!urbs[i]) continue; - + usb_kill_urb(urbs[i]); kfree(urbs[i]->transfer_buffer); usb_free_urb(urbs[i]); @@ -602,11 +602,11 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; - dev->n_audio_in = max(dev->spec.num_analog_audio_in, - dev->spec.num_digital_audio_in) / + dev->n_audio_in = max(dev->spec.num_analog_audio_in, + dev->spec.num_digital_audio_in) / CHANNELS_PER_STREAM; dev->n_audio_out = max(dev->spec.num_analog_audio_out, - dev->spec.num_digital_audio_out) / + dev->spec.num_digital_audio_out) / CHANNELS_PER_STREAM; dev->n_streams = max(dev->n_audio_in, dev->n_audio_out); @@ -619,7 +619,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) return -EINVAL; } - ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, + ret = snd_pcm_new(dev->chip.card, dev->product_name, 0, dev->n_audio_out, dev->n_audio_in, &dev->pcm); if (ret < 0) { @@ -632,7 +632,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) memset(dev->sub_playback, 0, sizeof(dev->sub_playback)); memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); - + memcpy(&dev->pcm_info, &snd_usb_caiaq_pcm_hardware, sizeof(snd_usb_caiaq_pcm_hardware)); @@ -651,9 +651,9 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) break; } - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_usb_caiaq_ops); - snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_usb_caiaq_ops); snd_pcm_lib_preallocate_pages_for_all(dev->pcm, @@ -662,7 +662,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) MAX_BUFFER_SIZE, MAX_BUFFER_SIZE); dev->data_cb_info = - kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, + kmalloc(sizeof(struct snd_usb_caiaq_cb_info) * N_URBS, GFP_KERNEL); if (!dev->data_cb_info) @@ -672,14 +672,14 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_cb_info[i].dev = dev; dev->data_cb_info[i].index = i; } - + dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); free_urbs(dev->data_urbs_in); return ret; } - + dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 515de1c..9be0f2e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -79,7 +79,7 @@ static struct usb_device_id snd_usb_id_table[] = { { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, .idVendor = USB_VID_NATIVEINSTRUMENTS, - .idProduct = USB_PID_RIGKONTROL2 + .idProduct = USB_PID_RIGKONTROL2 }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE, @@ -197,7 +197,7 @@ int snd_usb_caiaq_send_command(struct snd_usb_caiaqdev *dev, if (buffer && len > 0) memcpy(dev->ep1_out_buf+1, buffer, len); - + dev->ep1_out_buf[0] = command; return usb_bulk_msg(usb_dev, usb_sndbulkpipe(usb_dev, 1), dev->ep1_out_buf, len+1, &actual_len, 200); @@ -208,7 +208,7 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, { int ret; char tmp[5]; - + switch (rate) { case 44100: tmp[0] = SAMPLERATE_44100; break; case 48000: tmp[0] = SAMPLERATE_48000; break; @@ -237,12 +237,12 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (ret) return ret; - - if (!wait_event_timeout(dev->ep1_wait_queue, + + if (!wait_event_timeout(dev->ep1_wait_queue, dev->audio_parm_answer >= 0, HZ)) return -EPIPE; - - if (dev->audio_parm_answer != 1) + + if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); else dev->bpp = bpp; @@ -250,8 +250,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } -int snd_usb_caiaq_set_auto_msg (struct snd_usb_caiaqdev *dev, - int digital, int analog, int erp) +int snd_usb_caiaq_set_auto_msg(struct snd_usb_caiaqdev *dev, + int digital, int analog, int erp) { char tmp[3] = { digital, analog, erp }; return snd_usb_caiaq_send_command(dev, EP1_CMD_AUTO_MSG, @@ -262,7 +262,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) { int ret; char val[4]; - + /* device-specific startup specials */ switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): @@ -314,7 +314,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) dev->control_state, 1); break; } - + if (dev->spec.num_analog_audio_out + dev->spec.num_analog_audio_in + dev->spec.num_digital_audio_out + @@ -323,7 +323,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) if (ret < 0) log("Unable to set up audio system (ret=%d)\n", ret); } - + if (dev->spec.num_midi_in + dev->spec.num_midi_out > 0) { ret = snd_usb_caiaq_midi_init(dev); @@ -363,7 +363,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) if (devnum >= SNDRV_CARDS) return -ENODEV; - err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) return err; @@ -386,7 +386,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; int err, len; - + if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); return -EIO; @@ -395,19 +395,19 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_init_urb(&dev->ep1_in_urb); usb_init_urb(&dev->midi_out_urb); - usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, + usb_fill_bulk_urb(&dev->ep1_in_urb, usb_dev, usb_rcvbulkpipe(usb_dev, 0x1), - dev->ep1_in_buf, EP1_BUFSIZE, + dev->ep1_in_buf, EP1_BUFSIZE, usb_ep1_command_reply_dispatch, dev); - usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, + usb_fill_bulk_urb(&dev->midi_out_urb, usb_dev, usb_sndbulkpipe(usb_dev, 0x1), - dev->midi_out_buf, EP1_BUFSIZE, + dev->midi_out_buf, EP1_BUFSIZE, snd_usb_caiaq_midi_output_done, dev); - + init_waitqueue_head(&dev->ep1_wait_queue); init_waitqueue_head(&dev->prepare_wait_queue); - + if (usb_submit_urb(&dev->ep1_in_urb, GFP_KERNEL) != 0) return -EIO; @@ -420,10 +420,10 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) usb_string(usb_dev, usb_dev->descriptor.iManufacturer, dev->vendor_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iProduct, dev->product_name, CAIAQ_USB_STR_LEN); - + usb_string(usb_dev, usb_dev->descriptor.iSerialNumber, dev->serial, CAIAQ_USB_STR_LEN); @@ -431,7 +431,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) c = strchr(dev->serial, ' '); if (c) *c = '\0'; - + strcpy(card->driver, MODNAME); strcpy(card->shortname, dev->product_name); @@ -449,18 +449,18 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) return 0; } -static int __devinit snd_probe(struct usb_interface *intf, +static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - + ret = create_card(device, &card); - + if (ret < 0) return ret; - + usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); if (ret < 0) { @@ -468,7 +468,7 @@ static int __devinit snd_probe(struct usb_interface *intf, snd_card_free(card); return ret; } - + return 0; } @@ -489,10 +489,10 @@ static void snd_disconnect(struct usb_interface *intf) snd_usb_caiaq_input_free(dev); #endif snd_usb_caiaq_audio_free(dev); - + usb_kill_urb(&dev->ep1_in_urb); usb_kill_urb(&dev->midi_out_urb); - + snd_card_free(card); usb_reset_device(interface_to_usbdev(intf)); } diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 8fa8cd8..538e8c0 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -40,7 +40,7 @@ static void snd_usb_caiaq_midi_input_trigger(struct snd_rawmidi_substream *subst if (!dev) return; - + dev->midi_receive_substream = up ? substream : NULL; } @@ -64,18 +64,18 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, struct snd_rawmidi_substream *substream) { int len, ret; - + dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, EP1_BUFSIZE - 3); - + if (len <= 0) return; - + dev->midi_out_buf[2] = len; dev->midi_out_urb.transfer_buffer_length = len+3; - + ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," @@ -88,7 +88,7 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - + if (up) { dev->midi_out_substream = substream; if (!dev->midi_out_active) @@ -113,12 +113,12 @@ static struct snd_rawmidi_ops snd_usb_caiaq_midi_input = .trigger = snd_usb_caiaq_midi_input_trigger, }; -void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, +void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len) { if (!dev->midi_receive_substream) return; - + snd_rawmidi_receive(dev->midi_receive_substream, buf, len); } @@ -142,16 +142,16 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) if (device->spec.num_midi_out > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_usb_caiaq_midi_output); } if (device->spec.num_midi_in > 0) { rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; - snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_usb_caiaq_midi_input); } - + device->rmidi = rmidi; return 0; @@ -160,7 +160,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - + dev->midi_out_active = 0; if (urb->status != 0) return; -- 1.6.3.1

1 5

[alsa-devel] Testers wanted: New SB X-Fi driver
by gantonop＠cs.uoi.gr 30 May '09

30 May '09

I have an X-FI Platinum.Though I hear sound, it is periodically interrupted. More specifically every half a second an interrupt occurs producing a pause and a "tsik" :) which of course is annoying. So this has happened for every sound file I've tried to play.

1 0

[alsa-devel] (no subject)
by James Gardiner 30 May '09

30 May '09

Hi, someone recently submitted changes to the patch_sigmatel.c codec source. Whatever was done is breaking my patch for the HP dv4-1222nr laptop. I also strongly suspect that other code in the stac92hd71bxx function has been affected. There is no autodetect, there is no option for modprobe that will load that codec currently, to the best of my knowledge. I've tracked the difference to snapshot 20090529. Before that, I can get it to work, there and after refuses to work properly. Also, there was a recent change to the stac92xx_hp_check_power_status function: While the change looks good for its intended purpose of localizing board changes to the switch/case statements in the stac92hd71bxx function, the implementor reversed the if/then selector: if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) spec->gpio_data &= ~spec->gpio_led; /* white */ else spec->gpio_data |= spec->gpio_led; /* orange */ it should be: if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) spec->gpio_data |= spec->gpio_led; /* white */ else spec->gpio_data &= ~spec->gpio_led; /* orange */ The current implementation does not work-in fact sound is completely muted 100% of the time. I'll attempt to track down the 1st problem I mentioned, but I don't know if only patch_sigmatel.c was affected, or even the source of the problem. Thanks, James Gardiner

2 1

[alsa-devel] ALC883 features
by Michael Leies 30 May '09

30 May '09

Hi, I'm new here. I just bought a new computer from Dell with a Realtek ALC1200 (883?). After messing with alsa configurations and looking into the driver code a bit, I noted that a few features i'd like don't seem to be supported by the current driver. 1) Use of all 10 channels, only 8 are wired out to the 7.1 pcm device (forgive me if i lack the proper terminology for the moment). From the 883 datasheet, and comments in 1.0.20 code it seems the last 2 channels could be wired out as a separate PCM device. 2) Some more controls, specifically the ability to prevent the headphone jack from automuting the speakers. And if its possible to wire out the missing two channels then perhaps some more controls for the new channels. I was wondering if there someone who might point me in the right direction to get started with fixing up these things in the driver. Are my ambitions realistic, or am I barking up the tree of having to rewrite the whole HDA driver? Thanks, Michael

2 1

[alsa-devel] [PATCH v3 1/6] ARM: DaVinci: ASoC: Adds McASP clock support for TI DM646X processor
by Chaithrika U S 30 May '09

30 May '09

Adds McASP clock support for the two instances of mcasp (mcasp0,mcasp1). This patch is part of the audio support for dm646x series. 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/dm646x.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-davinci/dm646x.c b/arch/arm/mach-davinci/dm646x.c index f95b5e9..07c8d88 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -227,6 +227,18 @@ static struct clk gpio_clk = { .lpsc = DM646X_LPSC_GPIO, }; +static struct clk mcasp0_clk = { + .name = "mcasp0", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP0, +}; + +static struct clk mcasp1_clk = { + .name = "mcasp1", + .parent = &pll1_sysclk3, + .lpsc = DM646X_LPSC_McASP1, +}; + static struct clk aemif_clk = { .name = "aemif", .parent = &pll1_sysclk3, @@ -314,6 +326,8 @@ struct davinci_clk dm646x_clks[] = { CLK(NULL, "uart2", &uart2_clk), CLK("i2c_davinci.1", NULL, &i2c_clk), CLK(NULL, "gpio", &gpio_clk), + CLK(NULL, "mcasp0", &mcasp0_clk), + CLK(NULL, "mcasp1", &mcasp1_clk), CLK(NULL, "aemif", &aemif_clk), CLK("davinci_emac.1", NULL, &emac_clk), CLK(NULL, "pwm0", &pwm0_clk), -- 1.5.6

2 1

[alsa-devel] Add info in my sound-issue with the front Jack.
by Kristoffer Grundström 29 May '09

29 May '09

I added some possible info in my earlier bugreport about the front jack. Please visit this adress <https://qa.mandriva.com/show_bug.cgi?id=49048> and tell me if I've missed to add something. /Kristoffer Grundström P.S. I changed my emailadress since I got spammessages in my mailbox. D.S.

2 1

[alsa-devel] [PATCH v3 2/6] ARM: DaVinci: ASoC: Add the platform devices for ASP
by Chaithrika U S 29 May '09

29 May '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 phyisical 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 | 5 ++ arch/arm/mach-davinci/board-dm644x-evm.c | 5 ++ arch/arm/mach-davinci/board-dm646x-evm.c | 32 +++++++++++++++ arch/arm/mach-davinci/dm355.c | 41 ++++++++++++++++++- arch/arm/mach-davinci/dm644x.c | 36 ++++++++++++++++- arch/arm/mach-davinci/dm646x.c | 58 +++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 35 +++++++++++++++- arch/arm/mach-davinci/include/mach/dm355.h | 2 + arch/arm/mach-davinci/include/mach/dm644x.h | 2 + arch/arm/mach-davinci/include/mach/dm646x.h | 2 + 10 files changed, 213 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm355-evm.c b/arch/arm/mach-davinci/board-dm355-evm.c index f32e3d8..5bf1e13 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 evm_snd_platform_data dm355_evm_snd_data = { + .clk_name = "asp1", +}; + static int dm355evm_mmc_gpios = -EINVAL; static void dm355evm_mmcsd_gpios(unsigned gpio) @@ -278,6 +282,7 @@ static __init void dm355_evm_init(void) dm355_init_spi0(BIT(0), dm355_evm_spi_info, ARRAY_SIZE(dm355_evm_spi_info)); + dm355_init_asp(&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 e97216c..bdfbffe 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 evm_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 a55f45c..774497b 100644 --- a/arch/arm/mach-davinci/board-dm646x-evm.c +++ b/arch/arm/mach-davinci/board-dm646x-evm.c @@ -206,6 +206,37 @@ 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 evm_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), @@ -239,6 +270,7 @@ static __init void evm_init(void) evm_init_i2c(); davinci_serial_init(&uart_config); + dm646x_init_mcasp(dm646x_evm_snd_data); 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 737ba50..dca38e2 100644 --- a/arch/arm/mach-davinci/dm355.c +++ b/arch/arm/mach-davinci/dm355.c @@ -14,6 +14,7 @@ #include <linux/serial_8250.h> #include <linux/platform_device.h> #include <linux/dma-mapping.h> +#include <mach/asp.h> #include <linux/spi/spi.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), @@ -624,6 +625,32 @@ static struct platform_device dm355_edma_device = { .resource = edma_resources, }; +/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ +static struct resource dm355_asp_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_asp_device = { + .name = "davinci-asp", + .id = -1, + .num_resources = ARRAY_SIZE(dm355_asp_resources), + .resource = dm355_asp_resources, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm355_io_desc[] = { @@ -733,6 +760,16 @@ static struct davinci_soc_info davinci_soc_info_dm355 = { .sram_len = SZ_32K, }; +void __init dm355_init_asp(struct evm_snd_platform_data *pdata) +{ + /* we don't use ASP1 IRQs, or we'd need to mux them ... */ + davinci_cfg_reg(DM355_EVT8_ASP1_TX); + davinci_cfg_reg(DM355_EVT9_ASP1_RX); + + dm355_asp_device.dev.platform_data = pdata; + platform_device_register(&dm355_asp_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 7cea25b..8c6c23b 100644 --- a/arch/arm/mach-davinci/dm644x.c +++ b/arch/arm/mach-davinci/dm644x.c @@ -27,6 +27,7 @@ #include <mach/gpio.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), @@ -550,6 +551,32 @@ 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[] = { @@ -660,6 +687,13 @@ static struct davinci_soc_info davinci_soc_info_dm644x = { .sram_len = SZ_16K, }; +void __init dm644x_init_asp(struct evm_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 07c8d88..ad5f479 100644 --- a/arch/arm/mach-davinci/dm646x.c +++ b/arch/arm/mach-davinci/dm646x.c @@ -27,6 +27,7 @@ #include <mach/gpio.h> #include <mach/serial.h> #include <mach/common.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -587,6 +588,56 @@ static struct platform_device dm646x_edma_device = { .resource = edma_resources, }; +static struct resource dm646x_mcasp_resources[] = { + { + .name = "mcasp0", + .start = DAVINCI_DM646X_MCASP0_REG_BASE, + .end = DAVINCI_DM646X_MCASP0_REG_BASE + (SZ_1K << 1) - 1, + .flags = IORESOURCE_MEM, + }, + { + .name = "mcasp1", + .start = DAVINCI_DM646X_MCASP1_REG_BASE, + .end = DAVINCI_DM646X_MCASP1_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, + }, + /* 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_mcasp_device = { + .name = "davinci-mcasp", + .id = -1, + .num_resources = ARRAY_SIZE(dm646x_mcasp_resources), + .resource = dm646x_mcasp_resources, +}; + +static struct platform_device dm646x_dit_device = { + .name = "spdif-dit", + .id = -1, +}; + /*----------------------------------------------------------------------*/ static struct map_desc dm646x_io_desc[] = { @@ -697,6 +748,13 @@ static struct davinci_soc_info davinci_soc_info_dm646x = { .sram_len = SZ_32K, }; +void __init dm646x_init_mcasp(struct evm_snd_platform_data *pdata) +{ + dm646x_mcasp_device.dev.platform_data = pdata; + platform_device_register(&dm646x_mcasp_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..cffc55c 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 evm_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..b8f9c50 100644 --- a/arch/arm/mach-davinci/include/mach/dm355.h +++ b/arch/arm/mach-davinci/include/mach/dm355.h @@ -12,11 +12,13 @@ #define __ASM_ARCH_DM355_H #include <mach/hardware.h> +#include <mach/asp.h> 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_asp(struct evm_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..c508b5e 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 evm_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 7afc613..47b3423 100644 --- a/arch/arm/mach-davinci/include/mach/dm646x.h +++ b/arch/arm/mach-davinci/include/mach/dm646x.h @@ -14,6 +14,7 @@ #include <linux/platform_device.h> #include <mach/hardware.h> #include <mach/emac.h> +#include <mach/asp.h> #define DM646X_EMAC_BASE (0x01C80000) #define DM646X_EMAC_CNTRL_OFFSET (0x0000) @@ -23,5 +24,6 @@ #define DM646X_EMAC_CNTRL_RAM_SIZE (0x2000) void __init dm646x_init(void); +void __init dm646x_init_mcasp(struct evm_snd_platform_data *pdata); #endif /* __ASM_ARCH_DM646X_H */ -- 1.5.6

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[alsa-devel] Couple of alsa-tools patches from Gentoo
by Diego Elio 'Flameeyes' Pettenò 29 May '09

29 May '09

I already sent a link to the git repository where I pushed this last week, but given I'm actually going to remove it (need to reduce the number of repos on the server), I thought it would be easier just to send them as they are. One fixes build with glibc-2.10, the other fixes --as-needed in general. HTH, -- Diego E. Pettenò — Flameeyes

2 3

[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Using codec_regcache in twl4030_init_chip
by Peter Ujfalusi 29 May '09

29 May '09

Hello, The twl4030_init_chip used the twl4030_reg array directly to set the default register values. In other places the defaults has been modified in the codec->reg_cache (prime example is the Headset ramp delay configuration). Switch using the codec->reg_cache in the twl4030_init_chip function, so updated defaults will be actually used. --- Peter Ujfalusi (1): ASoC: TWL4030: Use reg_cache in twl4030_init_chip sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

2 2

Re: [alsa-devel] Acer Aspire One codec: Realtek ALC272 not found in ALSA-Configuration.txt Mic does not capture in alsamixer
by Takashi Iwai 29 May '09

29 May '09

At Sat, 23 May 2009 14:17:16 -0400, wirechief wrote: > > upon further review attempts have been un-successfull in getting the mic to work > with arecord -Ddefault:1 -f cd -vv /dev/null > also am finding this in dmesg: [ 11.684245] hda_codec: Unknown model > for ALC272, trying auto-probe from BIOS.. > my alsa http://www.alsa-project.org/db/?f=754f6ed51b0e45bf78662e50bde8f26546fb2bb4 Did you choose "Mic" as the input source? The default "model=auto" has no automatic mic-switching yet, so you'll have to choose the recording source manually. Takashi

1 0

[alsa-devel] [PATCH] ALSA: clean up the logic for building sequencer modules
by Michal Marek 29 May '09

29 May '09

Instead of mangling the CONFIG_* variables in the makefiles over and over, set a few helper variables in Kconfig. Signed-off-by: Michal Marek <mmarek(a)suse.cz> --- sound/core/Kconfig | 2 ++ sound/core/seq/Kconfig | 16 ++++++++++++++++ sound/core/seq/Makefile | 18 +++++------------- sound/drivers/opl3/Makefile | 10 +--------- sound/drivers/opl4/Makefile | 10 +--------- sound/isa/sb/Makefile | 10 +--------- sound/pci/emu10k1/Makefile | 10 +--------- sound/synth/Makefile | 12 ++---------- sound/synth/emux/Makefile | 12 ++---------- 9 files changed, 31 insertions(+), 69 deletions(-) create mode 100644 sound/core/seq/Kconfig diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 7bbdda0..6061fb5 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -205,3 +205,5 @@ config SND_PCM_XRUN_DEBUG config SND_VMASTER bool + +source "sound/core/seq/Kconfig" diff --git a/sound/core/seq/Kconfig b/sound/core/seq/Kconfig new file mode 100644 index 0000000..b851fd8 --- /dev/null +++ b/sound/core/seq/Kconfig @@ -0,0 +1,16 @@ +# define SND_XXX_SEQ to min(SND_SEQUENCER,SND_XXX) + +config SND_RAWMIDI_SEQ + def_tristate SND_SEQUENCER && SND_RAWMIDI + +config SND_OPL3_LIB_SEQ + def_tristate SND_SEQUENCER && SND_OPL3_LIB + +config SND_OPL4_LIB_SEQ + def_tristate SND_SEQUENCER && SND_OPL4_LIB + +config SND_SBAWE_SEQ + def_tristate SND_SEQUENCER && SND_SBAWE + +config SND_EMU10K1_SEQ + def_tristate SND_SEQUENCER && SND_EMU10K1 diff --git a/sound/core/seq/Makefile b/sound/core/seq/Makefile index 0695937..1bcb360 100644 --- a/sound/core/seq/Makefile +++ b/sound/core/seq/Makefile @@ -17,14 +17,6 @@ snd-seq-midi-event-objs := seq_midi_event.o snd-seq-dummy-objs := seq_dummy.o snd-seq-virmidi-objs := seq_virmidi.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_SEQUENCER) += snd-seq.o snd-seq-device.o ifeq ($(CONFIG_SND_SEQUENCER_OSS),y) obj-$(CONFIG_SND_SEQUENCER) += snd-seq-midi-event.o @@ -33,8 +25,8 @@ obj-$(CONFIG_SND_SEQ_DUMMY) += snd-seq-dummy.o # Toplevel Module Dependency obj-$(CONFIG_SND_VIRMIDI) += snd-seq-virmidi.o snd-seq-midi-event.o -obj-$(call sequencer,$(CONFIG_SND_RAWMIDI)) += snd-seq-midi.o snd-seq-midi-event.o -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-seq-midi-event.o snd-seq-midi-emul.o -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-seq-midi-emul.o snd-seq-virmidi.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-seq-midi-emul.o snd-seq-virmidi.o +obj-$(CONFIG_SND_RAWMIDI_SEQ) += snd-seq-midi.o snd-seq-midi-event.o +obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o +obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-seq-midi-event.o snd-seq-midi-emul.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-seq-midi-emul.o snd-seq-virmidi.o diff --git a/sound/drivers/opl3/Makefile b/sound/drivers/opl3/Makefile index 19767a6..7f2c2a1 100644 --- a/sound/drivers/opl3/Makefile +++ b/sound/drivers/opl3/Makefile @@ -7,14 +7,6 @@ snd-opl3-lib-objs := opl3_lib.o opl3_synth.o snd-opl3-synth-y := opl3_seq.o opl3_midi.o opl3_drums.o snd-opl3-synth-$(CONFIG_SND_SEQUENCER_OSS) += opl3_oss.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_OPL3_LIB) += snd-opl3-lib.o obj-$(CONFIG_SND_OPL4_LIB) += snd-opl3-lib.o -obj-$(call sequencer,$(CONFIG_SND_OPL3_LIB)) += snd-opl3-synth.o +obj-$(CONFIG_SND_OPL3_LIB_SEQ) += snd-opl3-synth.o diff --git a/sound/drivers/opl4/Makefile b/sound/drivers/opl4/Makefile index d178b39..b94009b 100644 --- a/sound/drivers/opl4/Makefile +++ b/sound/drivers/opl4/Makefile @@ -6,13 +6,5 @@ snd-opl4-lib-objs := opl4_lib.o opl4_mixer.o opl4_proc.o snd-opl4-synth-objs := opl4_seq.o opl4_synth.o yrw801.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - obj-$(CONFIG_SND_OPL4_LIB) += snd-opl4-lib.o -obj-$(call sequencer,$(CONFIG_SND_OPL4_LIB)) += snd-opl4-synth.o +obj-$(CONFIG_SND_OPL4_LIB_SEQ) += snd-opl4-synth.o diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index 1098a56..faeffce 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -13,14 +13,6 @@ snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o obj-$(CONFIG_SND_SB16_DSP) += snd-sb16-dsp.o @@ -33,4 +25,4 @@ ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o endif -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emu8000-synth.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emu8000-synth.o diff --git a/sound/pci/emu10k1/Makefile b/sound/pci/emu10k1/Makefile index cf2d563..fc5591e 100644 --- a/sound/pci/emu10k1/Makefile +++ b/sound/pci/emu10k1/Makefile @@ -9,15 +9,7 @@ snd-emu10k1-objs := emu10k1.o emu10k1_main.o \ snd-emu10k1-synth-objs := emu10k1_synth.o emu10k1_callback.o emu10k1_patch.o snd-emu10k1x-objs := emu10k1x.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-emu10k1.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emu10k1-synth.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emu10k1-synth.o obj-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o diff --git a/sound/synth/Makefile b/sound/synth/Makefile index e99fd76..11eb06a 100644 --- a/sound/synth/Makefile +++ b/sound/synth/Makefile @@ -5,16 +5,8 @@ snd-util-mem-objs := util_mem.o -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependency obj-$(CONFIG_SND_EMU10K1) += snd-util-mem.o obj-$(CONFIG_SND_TRIDENT) += snd-util-mem.o -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-util-mem.o -obj-$(call sequencer,$(CONFIG_SND)) += emux/ +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-util-mem.o +obj-$(CONFIG_SND_SEQUENCER) += emux/ diff --git a/sound/synth/emux/Makefile b/sound/synth/emux/Makefile index b690352..328594e 100644 --- a/sound/synth/emux/Makefile +++ b/sound/synth/emux/Makefile @@ -7,14 +7,6 @@ snd-emux-synth-objs := emux.o emux_synth.o emux_seq.o emux_nrpn.o \ emux_effect.o emux_proc.o emux_hwdep.o soundfont.o \ $(if $(CONFIG_SND_SEQUENCER_OSS),emux_oss.o) -# -# this function returns: -# "m" - CONFIG_SND_SEQUENCER is m -# <empty string> - CONFIG_SND_SEQUENCER is undefined -# otherwise parameter #1 value -# -sequencer = $(if $(subst y,,$(CONFIG_SND_SEQUENCER)),$(if $(1),m),$(if $(CONFIG_SND_SEQUENCER),$(1))) - # Toplevel Module Dependencies -obj-$(call sequencer,$(CONFIG_SND_SBAWE)) += snd-emux-synth.o -obj-$(call sequencer,$(CONFIG_SND_EMU10K1)) += snd-emux-synth.o +obj-$(CONFIG_SND_SBAWE_SEQ) += snd-emux-synth.o +obj-$(CONFIG_SND_EMU10K1_SEQ) += snd-emux-synth.o -- 1.6.3

2 2

[alsa-devel] Increasing maximum sampling frequency?
by Cameron Stone 29 May '09

29 May '09

I've got a USB microphone application that I want to sample at 256000 8 bit samples per second so I can listen to bats. I know that my device can get close to that rate, but alsa seems to restrict sampling rates to 192kHz. Is it possible to push the maximum sampling frequency higher? If so, where should I start looking? I tried hacking aplay.c so arecord would accept command line requests up to 256k, but it seems the alsa driver is also limiting the frequency to a maximum of 192k. However, the files generated by arecord *do* have the right number of samples in them. $ arecord -v -Dplughw:1,0 test.wav -d 1 -r 256 Recording WAVE 'test.wav' : Unsigned 8 bit, Rate 256000 Hz, Mono Plug PCM: Linear conversion PCM (S8) Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 256000 exact rate : 256000 (256000/1) msbits : 8 buffer_size : 128000 period_size : 32000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 32000 period_event : 0 start_threshold : 1 stop_threshold : 128000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 1 '8-Mic AVR Adaptor' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S8 subformat : STD channels : 1 rate : 256000 exact rate : 256000 (256000/1) msbits : 8 buffer_size : 128000 period_size : 32000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 32000 period_event : 0 start_threshold : 1 stop_threshold : 128000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 $ aplay -r 256 test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 256000 Hz, Mono Warning: rate is not accurate (requested = 256000Hz, got = 192000Hz) please, try the plug plugin (-Dplug:default) $ ls -l test.wav -rw-r--r-- 1 camerons camerons 256044 2009-05-28 11:29 test.wav $ file test.wav test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit, mono 256000 Hz $

5 7

Re: [alsa-devel] Increasing maximum sampling frequency?
by Cameron Stone 29 May '09

29 May '09

The A/D chip is an Analog Devices 7928. It can do 2 mega-samples per second, multiplexed across 8 inputs. The USB audio interface is provided by my firmware code, assisted by the AVR AT90USB647 micro which has some low-level USB support. HTH. Cameron. Alex Austin wrote: > By Codec, I think he's asking about the A/D chip. For example, the TI > PCM2902E is a stereo USB audio codec. > > - Alex Austin > (651) 238-9273 > > "...and then I visited Wikipedia ...and the next 8 hours are a blur." > > > On Thu, May 28, 2009 at 8:30 AM, Cameron Stone > <camerons.lists(a)cse.unsw.edu.au <mailto:camerons.lists@cse.unsw.edu.au>> > wrote: > > Daniel Mack wrote: > > On Thu, May 28, 2009 at 11:47:34AM +1000, Cameron Stone wrote: > >> I've got a USB microphone application that I want to sample at > 256000 8 > >> bit samples per second so I can listen to bats. I know that my device > >> can get close to that rate, but alsa seems to restrict sampling > rates to > >> 192kHz. > > > > Just out of curiosity - which microphone and what kind of codec > are you > > using for such frequencies? > > I don't know which microphone but I can find out if you're interested. I > do know that its sensitivity starts at 10kHz, if that helps. > > There is no codec used. I'm using Type I (uncompressed PCM) because it's > being powered by an AVR running at 16MHz and there's not enough clock > cycles to encode the samples. > > Cameron. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org <mailto:Alsa-devel@alsa-project.org> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

1 0

[alsa-devel] [PATCH V3 0/4] mpc5200 audio rework for AC97
by Jon Smirl 28 May '09

28 May '09

The following series implements audio support for the mpc5200. It adds an AC97 driver and STAC9766 codec driver. Board support for the Efika and Phytec pcm030 are also included. I've tried to implement the feedback received on the previous two versions. based on commit 0bc53a67ac831ec84f730a657dbcadd80a589ef5 on broonie/for-2.6.31 at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git --- Jon Smirl (4): Fabric bindings for STAC9766 on the Efika Support for AC97 on Phytec pmc030 base board. AC97 driver for mpc5200 Main rewite of the mpc5200 audio DMA code sound/soc/fsl/Kconfig | 27 ++ sound/soc/fsl/Makefile | 5 sound/soc/fsl/efika-audio-fabric.c | 95 +++++++ sound/soc/fsl/mpc5200_dma.c | 504 +++++++++++++++++++++++------------ sound/soc/fsl/mpc5200_dma.h | 33 +- sound/soc/fsl/mpc5200_psc_ac97.c | 392 +++++++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_ac97.h | 15 + sound/soc/fsl/mpc5200_psc_i2s.c | 247 +++-------------- sound/soc/fsl/mpc5200_psc_i2s.h | 12 + sound/soc/fsl/pcm030-audio-fabric.c | 95 +++++++ 10 files changed, 1036 insertions(+), 389 deletions(-) create mode 100644 sound/soc/fsl/efika-audio-fabric.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h create mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h create mode 100644 sound/soc/fsl/pcm030-audio-fabric.c -- Jon Smirl jonsmirl(a)gmail.com

4 21

[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 28 May '09

28 May '09

The following changes since commit 08d15f034e94251606479d7ca9070994c2e2fcf0: Mark Brown (1): ASoC: Switch FSL SSI DAI over to symmetric_rates 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 Chaithrika U S (1): ASoC: Add dummy S/PDIF codec support Jon Smirl (1): ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout() Mark Brown (1): ASoC: Initialise dev for the dummy S/PDIF DAI Roel Kluin (1): ASoC: correct print specifiers for unsigneds sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/spdif_transciever.c | 71 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 ++++++++ sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/uda134x.c | 4 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 4 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 6 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 30 +++++++-------- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 4 +- sound/soc/sh/ssi.c | 2 +- 19 files changed, 130 insertions(+), 38 deletions(-) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h

2 1

[alsa-devel] [PATCH v3 4/6] ARM: DaVinci: ASoC: Add dummy codec support
by Chaithrika U S 28 May '09

28 May '09

McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed. This patch provides stub codec that can be used in these configurations. On DM646x EVM the McASP1 is connected to the S/PDIF out. 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 - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/spdif_transciever.c | 69 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/spdif_transciever.h | 17 ++++++++ 4 files changed, 92 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/spdif_transciever.c create mode 100644 sound/soc/codecs/spdif_transciever.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb07d9b..bbc97fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 + select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C @@ -91,6 +92,9 @@ config SND_SOC_L3 config SND_SOC_PCM3008 tristate +config SND_SOC_SPDIF + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 46c007c..8b75305 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o @@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c new file mode 100644 index 0000000..118e976 --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.c @@ -0,0 +1,69 @@ +/* + * ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIT (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. TI DaVinci Audio controller uses this driver. + * + * Author: Steve Chen, <schen(a)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/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> +#include <sound/pcm.h> + +#define STUB_RATES SNDRV_PCM_RATE_8000_96000 +#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE + + +struct snd_soc_dai dit_stub_dai = { + .name = "DIT", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dit_probe(struct platform_device *pdev) +{ + return snd_soc_register_dai(&dit_stub_dai); +} + +static int spdif_dit_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&dit_stub_dai); + return 0; +} + +static struct platform_driver spdif_dit_driver = { + .probe = spdif_dit_probe, + .remove = spdif_dit_remove, + .driver = { + .name = "spdif-dit", + .owner = THIS_MODULE, + }, +}; + +static int __init dit_modinit(void) +{ + return platform_driver_register(&spdif_dit_driver); +} + +static void __exit dit_exit(void) +{ + platform_driver_unregister(&spdif_dit_driver); +} + +module_init(dit_modinit); +module_exit(dit_exit); + diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h new file mode 100644 index 0000000..296f2eb --- /dev/null +++ b/sound/soc/codecs/spdif_transciever.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.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 + * published by the Free Software Foundation. + */ + +#ifndef CODEC_STUBS_H +#define CODEC_STUBS_H + +extern struct snd_soc_dai dit_stub_dai; + +#endif /* CODEC_STUBS_H */ -- 1.5.6

2 1

[alsa-devel] [PATCH v3 6/6] ARM: DaVinci: ASoC: Add machine driver support for DM646x
by Chaithrika U S 28 May '09

28 May '09

This patch does the following: (1) Add support for the DM646x machine (2)Modifications required to introduce the platform driver model to get platform data for all the machines including dm355 and dm644x. 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/davinci-evm.c | 103 ++++++++++++++++++-------------------- 1 files changed, 49 insertions(+), 54 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 58fd1cb..d66fbd3 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -25,11 +25,13 @@ #include <mach/asp.h> #include <mach/edma.h> #include <mach/mux.h> +#include <mach/cpu.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" - +#include "davinci-i2s-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -43,7 +45,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -144,6 +146,24 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_iis_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai = &davinci_iis_mcasp_dai[DAVINCI_MCASP_DIT_DAI], + .codec_dai = &dit_stub_dai, + .ops = &evm_ops, + }, +}; + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,12 +172,26 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = ARRAY_SIZE(dm6467_evm_dai), +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x1b, }; +/* dm6467 evm audio private data */ +static struct aic3x_setup_data dm6467_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -165,60 +199,30 @@ static struct snd_soc_device evm_snd_devdata = { .codec_data = &evm_aic3x_setup, }; -/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_ASP0_BASE, - .end = DAVINCI_ASP0_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP0_TX, - .rx_dma_ch = DAVINCI_DMA_ASP0_RX, -}; - -/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ -static struct resource dm335evm_snd_resources[] = { - { - .start = DAVINCI_ASP1_BASE, - .end = DAVINCI_ASP1_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct evm_snd_platform_data dm335evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP1_TX, - .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &dm6467_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { - struct resource *resources; - unsigned num_resources; - struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; if (machine_is_davinci_evm()) { - davinci_cfg_reg(DM644X_MCBSP); - - resources = evm_snd_resources; - num_resources = ARRAY_SIZE(evm_snd_resources); - data = &evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { - /* we don't use ASP1 IRQs, or we'd need to mux them ... */ - davinci_cfg_reg(DM355_EVT8_ASP1_TX); - davinci_cfg_reg(DM355_EVT9_ASP1_RX); - - resources = dm335evm_snd_resources; - num_resources = ARRAY_SIZE(dm335evm_snd_resources); - data = &dm335evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; } else return -EINVAL; @@ -226,17 +230,8 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(*data)); - - ret = platform_device_add_resources(evm_snd_device, resources, - num_resources); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); -- 1.5.6

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[alsa-devel] [PATCH v3 3/6] ARM: DaVinci: ASoC: Introduce platform driver model for dm644x, dm355
by Chaithrika U S 28 May '09

28 May '09

Introduce the platform driver model to get platform data for dm355 and dm644x. Register platform driver and acquire the resources in the probe function Since the platform specific code had been moved from machine driver to dm<soc>.c 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 - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git sound/soc/davinci/davinci-i2s.c | 115 ++++++++++++++++++++++----------------- 1 files changed, 66 insertions(+), 49 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b1ea52f..dec7611 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -436,15 +436,39 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct evm_snd_platform_data *pdata = pdev->dev.platform_data;; struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; + struct resource *mem, *ioarea, *res; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -466,9 +490,7 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - - dev->clk = clk_get(&pdev->dev, NULL); + dev->clk = clk_get(&pdev->dev, pdata->clk_name); if (IS_ERR(dev->clk)) { ret = -ENODEV; goto err_free_mem; @@ -476,18 +498,35 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; + return 0; err_free_mem: @@ -498,62 +537,40 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int davinci_i2s_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); -} -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + return 0; +} -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asp", + .owner = THIS_MODULE, + }, }; -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); - static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_mcbsp_driver); } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_mcbsp_driver); } module_exit(davinci_i2s_exit); -- 1.5.6

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[alsa-devel] [PATCH v3 0/6] ARM: DaVinci: Audio support for DM646x
by Chaithrika U S 28 May '09

28 May '09

From: Chaithrika U S <chaithrika(a)ti.com> Add audio support for DM646x Signed-off-by: Naresh Medisetty <naresh(a)ti.com> Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch set adds audio support for DM646x This updated version has included most of the review comments received for the previous version of the patch set. This series is dependent on the following patches submitted to the DaVinci mailing list: [PATCH v2 1/2] ARM: DaVinci: EDMA: Move queue related mappings to dm<soc>.c [PATCH v2 2/2] ARM: DaVinci: EDMA: Enable EDMA support for dm646x Patch 1: Adds McASP clock support for the two instances of mcasp(mcasp0,mcasp1) This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 2: Registers the platform devices for ASP on dm355, dm644x and dm646x This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git Patch 3: Introduces platform driver model for dm644x and dm355 Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 4: Add dummy codec support Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 5: Adds driver support for the two instances of McASP on TI's DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Patch 6: Add machine driver support for DM646x Applies against ALSA mainline - that's the topic/asoc branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git These patches are based on the work done by Steve Chen <schen(a)mvista.com> and Pavel Kiryukhin <pkiryukhin(a)ru.mvista.com> on Monstavista 2.6.18 kernel.

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[alsa-devel] [PATCH] Compaq Presario CQ60 patching for Conexant
by Tony Vroon 28 May '09

28 May '09

A docking mic control is shown by default. The Compaq Presario CQ60 laptop has no docking connector, so designate it as a CXT5051_HP model. This makes the phantom mixer slider disappear. Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 56ce19e..4fcbe21 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1848,6 +1848,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), + SND_PCI_QUIRK(0x103c, 0x360b, "Compaq Presario CQ60", CXT5051_HP), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),

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[alsa-devel] Testers wanted: New SB X-Fi driver
by jimi o'beirne 28 May '09

28 May '09

Takashi Thanks so much for making this driver! Unfortunately, I'm a bit of a newbie, how would I actually install the driver? I am using Ubuntu Jaunty with a SB X-Fi card. Thanks for your help, Jim

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#529940: lib32asound2: symbol versioning changed in an incompatible way
by Elimar Riesebieter 28 May '09

28 May '09

tags 529940 pending forwarded 529940 alsa-devel(a)alsa-project.org thanks * Adeodato Simó [090527 13:53 +0200] > severity 529940 serious > retitle 529940 lib32asound2: symbol versioning changed in an incompatible way (ALSA_0.9.0rc4/8 dropped) > thanks > > Hello, > > it seems that between 1.0.19-1 and 1.0.20-1 the list of symbols in > lib32asound2 changed. More precisely, the list of symbols remained but > their associated versioning changed: all symbols associated with > ALSA_0.9.0rc4 and ALSA_0.9.0rc8 were changed to live under ALSA_0.9 > instead, making applications compiled against <= 1.0.19 fail to start > due to missing symbols. > > Note that this problem affects the 32-bit library in amd64, but not the > native libraries nor in i386 nor amd64. Don't we have to verify it at ppc64 as well? CC'd to debian-ppc. > > Also note that we've started having applications like WINE 1.1.19 in > experimental rebuilt against 1.0.20, and which don't work against > alsa-lib 1.0.19 either. And we'll have no easy way to detect these, > since shlibs are not bumped. > > Will you be able to upload fixed packages in a timely manner? According to [0] I created svn version 2249 on alioth [1]. I've tested it on amd64, were rc4 symbols appeared by checking via readelf. But maybe alsa-devel rechecks the buildsystem of alsa-libs to avoid that dirty hacks? --with-versioned appended to ./configure opts didn't help. [0] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016083.html [1] http://svn.debian.org/wsvn/pkg-alsa/?op=comp&compare[]=%2F@2248&compare[]=%… Jordi, please recheck and upload. Index: debian/rules =================================================================== --- debian/rules (Revision 2248) +++ debian/rules (Arbeitskopie) @@ -39,7 +39,7 @@ endif endif -CFLAGS = -g -Wall +CFLAGS = -g -Wall -DUSE_VERSIONED_SYMBOLS ifneq (,$(findstring noopt,$(DEB_BUILD_OPTIONS))) CFLAGS += -O0 -- "Talking much about oneself can also be a means to conceal oneself." -Friedrich Nietzsche

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