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

[alsa-devel] Playback sound is only audible starting at 50% on Dell Inspiron 1420 (Ubuntu 32-bit)
by komputes 02 Jun '09

02 Jun '09

Hello ALSA Developers, I would like to know if anyone can help me correct the following issue. Playback sound is only audible starting at 50% on Dell Inspiron 1420 running Ubuntu 9.04 Desktop x86. This is true with 9.04, 8.10, 8.04 (all x86) but does not affect Ubuntu 9.04 AMD64 on which sound is audible at ~4%. Both of these volume levels refer to the "Master" level, all other playback levels are at 100% (even the ones that are invisible/disabled by default). I hang out on #alsa where I was proposed to start alsa with the options "options snd-hda-intel model=dell-3stack" then "options snd-hda-intel model=3stack" none of which corrected this 50% volume issue. This issue as well as all technical details can be found here (includes alsa-info output): https://bugs.edge.launchpad.net/ubuntu/+source/alsa-driver/+bug/382488 Thank you for your time, -komputes

2 1

[alsa-devel] [PATCH 1/2] ALSA: snd_usb_caiaq: give better shortname
by Daniel Mack 02 Jun '09

02 Jun '09

If not passed as module option, provide an own card ID with the newly introduced snd_set_card_id() call. This will prevent ALSA from calling choose_default_name() which only takes the last part of a name containing whitespaces. This for example caused 'Audio 4 DJ' to be shortened to 'DJ', which was not very descriptive. The implementation now takes the short name and removes all whitespaces from it which is much nicer. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 21 +++++++++++++++++++-- 1 files changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 15052e4..43bd0db 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -382,10 +382,10 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) static int __devinit init_card(struct snd_usb_caiaqdev *dev) { - char usbpath[32]; + char *c, usbpath[32]; struct usb_device *usb_dev = dev->chip.dev; struct snd_card *card = dev->chip.card; - int err; + int err, len; if (usb_set_interface(usb_dev, 0, 1) != 0) { log("can't set alt interface.\n"); @@ -427,6 +427,23 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + /* if the id was not passed as module option, fill it with a shortened + * version of the product string which does not contain any + * whitespaces */ + + if (*card->id == '\0') { + char id[sizeof(card->id)]; + + memset(id, 0, sizeof(id)); + + for (c = card->shortname, len = 0; + *c && len < sizeof(card->id); c++) + if (*c != ' ') + id[len++] = *c; + + snd_card_set_id(card, id); + } + usb_make_path(usb_dev, usbpath, sizeof(usbpath)); snprintf(card->longname, sizeof(card->longname), "%s %s (%s)", -- 1.6.3.1

2 4

[alsa-devel] [PATCH 1/5] ALSA: snd_usb_caiaq: clean whitespaces
by Daniel Mack 02 Jun '09

02 Jun '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

3 23

Re: [alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

2009/6/2 Mark Brown <broonie(a)opensource.wolfsonmicro.com> > On Tue, Jun 02, 2009 at 03:24:11PM +0530, Naveen Krishna Ch wrote: > > > I am done with the entire back porting of ASoC. > > > > I can see the dapm widgets getting ON/OFF and codec registers being > changed. > > > > But, there is no audio out. Am i missing somthing > > > > A direct aplay should play the audio on the headset, right? > > I am not familiar with the TWL4030, you should ask these questions on > the list. I hit reply (Instead of replyall) Sorry, > As a first step I would check that there is data being > transferred over the digital link to the CODEC, then look at the > audio routing to ensure that the signal is routed to the output. Thanks for suggestions.. -- Cheers (: Naveen Krishna Ch :)

2 1

[alsa-devel] Palm27x-asoc: fix it to work without jack gpio specified
by Marek Vasut 02 Jun '09

02 Jun '09

Hi, the patch follows. The problem was with palmz72 which doesn't have the jack detection.

2 3

[alsa-devel] ASoC: TWL4030 driver support multi channel playback
by Naveen Krishna Ch 02 Jun '09

02 Jun '09

Hi Peter, I am using linux 2.6.28 kernel, there is a requirement to support multi (4) channel play back . I guess TWL4030 ASoC driver now has full fledged support for both the DAC1(TX1) and DAC2(TX2) paths in option 1 of codec operation. I am looking forward to back port the ASoC stuff drivers from 2.6.30 rc7 to 2.6.28. Is there any better way of meeting the requirement with out back porting the stuff Suggestions please. Sorry for the english,, -- Cheers (: Naveen Krishna Ch :)

3 4

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..8503a01 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,51 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[PATCH] ALSA: introduce snd_card_make_id_unique()
by Daniel Mack 02 Jun '09

02 Jun '09

Sound card drivers may choose their own way to provide a nice sound card ID, especially when the default behaviour taken by choose_default_name() is not appropriate. ALSA's IDs must be unique, so there is need for some core logic to ensure that. This patch adds the function snd_card_make_id_unique() to the ALSA core which lowlevel drivers can call to make sure the computed string isn't already taken. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/core.h | 1 + sound/core/init.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 54 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3dea798..01ea816 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -319,6 +319,7 @@ int snd_card_info_done(void); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); +void snd_card_make_id_unique(struct snd_card *card); #ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) diff --git a/sound/core/init.c b/sound/core/init.c index fd56afe..bce2982 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -547,6 +547,59 @@ static void choose_default_id(struct snd_card *card) } } +static int +card_id_is_ambiguous(struct snd_card *card, const char *id) +{ + int i; + + for (i = 0; i < snd_ecards_limit; i++) { + /* only run for exisiting cards but not for the + * one in question */ + if (i == card->number || !snd_cards[i]) + continue; + + if (strcmp(snd_cards[i]->id, id) == 0) + return 1; + } + + return 0; +} + +void +snd_card_make_id_unique(struct snd_card *card) +{ + int count = 1; + char id[sizeof(card->id)]; + + if (*card->id == '\0') + return; + + /* if there's a reserved word in the card's chosen id, reset it + * to empty. choose_default_id() will then take care to generate + * an id for the card at a later point */ + if (!snd_info_check_reserved_words(card->id)) { + *card->id = '\0'; + return; + } + + strlcpy(id, card->id, sizeof(id)); + + while (card_id_is_ambiguous(card, id)) { + /* As we need to add some characters to the id, make + * sure there's enough room for it */ + int maxlen = sizeof(card->id) - (2 + (count / 10)); + + if (strlen(card->id) > maxlen) + card->id[maxlen] = '\0'; + + snprintf(id, sizeof(id), "%s_%d", card->id, count); + count++; + } + + strlcpy(card->id, id, sizeof(id)); +} +EXPORT_SYMBOL_GPL(snd_card_make_id_unique); + #ifndef CONFIG_SYSFS_DEPRECATED static ssize_t card_id_show_attr(struct device *dev, -- 1.6.3.1

1 0

[alsa-devel] [PATCH] Acer Aspire 8930G support
by Hector Martin 02 Jun '09

02 Jun '09

Getting this one to work has annoyed me to no end. Short story: this laptop has 5.1 built-in speakers which you *really* want to use (the not-so-"sub" woofer is what makes the audio above average for a laptop), so 6-channel support is important (plus a decent asound.conf to upmix stereo). It also has the 3 typical jacks that ought to have a selectable mode. And it's based on ALC889, which sucks. Rationale/explanations: The const_channel_count stuff was added because, for a laptop like this, you always have 6 channels available (internal speakers) but still need to set the mode for the 3 external jacks. Therefore, the device always needs to be in 6-channel mode but there still needs to be a mixer control for the jack mode. You could use line/mic-in at the same time as the 6 internal speakers, for example. You might be tempted to make it even smarter by dynamically switching the max channel count when headphones are plugged in (therefore muting the internal speakers and reducing the physical channel count to the jack channel mode), but as a user I consider this to be harmful because I want the audio to blow up to 6 channels / upmixed as soon as I unplug the headphones, and having opened the device while in 2-channel mode would prevent this from working (and always making 6-channel mode available doesn't do any harm). The hardware needs EAPD turned on and the DACs routed to the internal speaker pins, so the patch adds those verbs. The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT work by default, at least here. I wasted much time trying to talk to Realtek/pshou about this, but they just kept sending me useless updates to patch_realtek.c that did nothing relevant. In the end I gave up and brute forced the issue by trying to flip every bit in the proprietary coefficient registers, and eventually found the two magic registers that need to be cleared to enable all DACs. I have only heard Acer users complain, but that might be because ALC889 is pretty new and using 5.1 (and noticing the missing center/lfe channels) might not be that common. If this is a generalized issue with all ALC889 systems then those verbs should probably be moved to a common verb array. The internal mic is untested and probably doesn't work. These settings will probably work for other Acer Gemstone laptops with the same 5.1 speaker config. When identified, those should be added to the PCI subsystem ID list. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek_orig.c 2009-06-02 07:48:47.000000000 +0200 +++ alsa-driver-1.0.20.8.g6254b.219.g5b2b1/alsa-kernel/pci/hda/patch_realtek.c 2009-06-02 08:32:29.000000000 +0200 @@ -223,6 +223,7 @@ ALC883_ACER, ALC883_ACER_ASPIRE, ALC888_ACER_ASPIRE_4930G, + ALC888_ACER_ASPIRE_8930G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -313,6 +314,8 @@ const struct hda_channel_mode *channel_mode; int num_channel_mode; int need_dac_fix; + int const_channel_count; + int ext_channel_count; /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ @@ -368,6 +371,7 @@ unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; int need_dac_fix; + int const_channel_count; unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); @@ -462,7 +466,7 @@ struct alc_spec *spec = codec->spec; return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - spec->multiout.max_channels); + spec->ext_channel_count); } static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, @@ -472,9 +476,12 @@ struct alc_spec *spec = codec->spec; int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; + &spec->ext_channel_count); + if(err >= 0 && spec->const_channel_count == 0) { + spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; + } return err; } @@ -854,8 +861,13 @@ spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; + spec->const_channel_count = preset->const_channel_count; - spec->multiout.max_channels = spec->channel_mode[0].channels; + if(preset->const_channel_count == 0) + spec->multiout.max_channels = preset->const_channel_count; + else + spec->multiout.max_channels = spec->channel_mode[0].channels; + spec->ext_channel_count = spec->channel_mode[0].channels; spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; @@ -1456,6 +1468,44 @@ { } }; +/* + * ALC888 Acer Aspire 8930G model + */ + +static struct hda_verb alc888_acer_aspire_8930g_verbs[] = { +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, +/* Enable unsolicited event for HP jack */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, +/* Connect Internal Front to Front */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Internal Rear to Rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect Internal CLFE to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Connect HP out to Front */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Enable all DACs */ + {0x20, AC_VERB_SET_COEF_INDEX, 0x03}, // DAC DISABLE/MUTE 1? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // bits 1-5, when set, disable DAC nids 0x02-0x06 apparently. Init value 0x38 + {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, // DAC DISABLE/MUTE 2? + {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, // some bit here disables the other DACs. Init value 0x4900 +/* Enable amplifiers */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + { } +}; + static struct hda_input_mux alc888_2_capture_sources[2] = { /* Front mic only available on one ADC */ { @@ -1508,6 +1558,17 @@ alc_automute_amp(codec); } +static void alc888_acer_aspire_8930g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x1b; + alc_automute_amp(codec); +} + /* * ALC880 3-stack model * @@ -8708,6 +8769,7 @@ [ALC883_ACER] = "acer", [ALC883_ACER_ASPIRE] = "acer-aspire", [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", + [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -8746,6 +8808,8 @@ ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0145, "Acer Aspire 8930G", + ALC888_ACER_ASPIRE_8930G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ @@ -8959,6 +9023,27 @@ .unsol_event = alc_automute_amp_unsol_event, .init_hook = alc888_acer_aspire_4930g_init_hook, }, + [ALC888_ACER_ASPIRE_8930G] = { + .mixers = { alc888_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_8930g_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .const_channel_count = 6, + .num_mux_defs = + ARRAY_SIZE(alc888_2_capture_sources), + .input_mux = alc888_2_capture_sources, + .unsol_event = alc_automute_amp_unsol_event, + .init_hook = alc888_acer_aspire_8930g_init_hook, + }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, alc883_chmode_mixer },

2 5

[alsa-devel] SPDIF/AES input sample rate handling?
by Robert Hancock 02 Jun '09

02 Jun '09

A question about how ALSA drivers should handle SPDIF or AES digital audio inputs: I may be eventually be writing an ALSA driver for an audio card which has multiple AES digital audio inputs. The card would support multiple sample rates on input with the provided input data being fed directly to software with no sample rate conversion. This raises the question of how to handle requests for a particular sample rate. If the user app requests recording at a sample rate of say 32KHz and we say "OK", and then they proceed to connect an input running at 44.1KHz sampling rate, the card will produce and the app will receive samples at a faster than expected rate. Presumably this is something the app just has to deal with, but will anything in the ALSA core/library puke because of it? Or is there a better way to deal with this kind of situation?

2 1