April 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC PATCH] Inverted internal mic
by David Henningsson 25 Jun '12

25 Jun '12

Hi, One of the more common problems on laptop machines, is that the internal mic provides a stereo signal but with one channel phase inverted, or differential output. This means problems for applications recording two channels but later merging them into one, leaving them with zero or near-zero output. There are various ways we can work around this in both the kernel, alsa-lib and PulseAudio layers. It's a matter of picking the right one. I'm leaning towards trying to fix it in the kernel's codec drivers, because 1) we already have quirking infrastructure there 2) we already have some working quirks already in that layer 3) it would benefit other sound applications that use ALSA directly. The downside to that is really that we're silencing out one channel for everyone, leading to no application being able to use both channels, even if they would implement some kind of "auto-detect-and-reverse-one-channel" functionality. For the most part, this has been some Acer Aspire machines running Realtek ALC268 / ALC269 / ALC271X / ALC272X, and for two of these we have proc coefs we can set, but for the other two these proc coefs, if they exist, are unknown to us. Recently I came across a Conexant as well, and I decided to write a patch for it, that would take the approach that the internal mic is forced mono on the kcontrol, and make sure the right channel is always muted. The patch is verified by the reporter to fix this problem. It could use some perfection though - it would make sense to to the same to the internal mic boost as well, and the strcmp('Internal Mic') call could maybe be turned into something more elegant. But before going ahead with that, I'd like to hear your opinion on the matter, if you agree that this is a good approach to the problem? References: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/903853/+attachme… https://bugs.launchpad.net/bugs/903853 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

5 38

Re: [alsa-devel] [PATCH v3] sound/soc/lapis: add platform driver for ML7213
by Mark Brown 13 Jun '12

13 Jun '12

On Tue, Mar 06, 2012 at 02:48:05PM +0900, Tomoya MORINAGA wrote: > 2012年3月3日1:39 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > I just merged Lars-Peter's dmaengine library code which has been on the > > list for a week or so - this should be updated to use that next time > > it's posted. That should save a lot of code from the driver and make > > sure it's following best practices for dmaengine use. > Sorry, we can't use this library. This sort of response really isn't on. Do you have some reason for saying this? Flat out refusing to do things with no reason is not useful. > >> +static struct ioh_i2s_data *i2s_data; > >> +static struct ioh_i2s_dma dmadata[MAX_I2S_CH]; > > Why are these needed, aren't they dynamically allocated by the driver? > You mean ASoC driver can't use global variable ? In general no Linux driver should be using a global variable except for things like this. > >> + case SNDRV_PCM_FORMAT_S32_LE: > >> + byte = 24; > >> + break; > > That looks wrong... are you sure you don't support S24_LE or something? > This is correct. > Because maximum transmit size of ML7213's I2S hw is 24bit. Note we often lay out 24 bit audio in 32 bit blocks. > >> +static struct platform_driver ioh_i2s_driver_plat = { > > > >> +static struct platform_driver ioh_dai_driver_plat = { > > > >> +static int ioh_i2s_pci_probe(struct pci_dev *pdev, > >> + const struct pci_device_id *id) > > Why are you creating these platform devices? I don't understand the > > function they serve. The code handling them looks to have quite a few > > problems but I'm not clear they should be there in the first place. > if these platform devices aren't used, device detection doesn't work correctly. > So, I added these. You've not actually mentioned the problem you were seeing... > >> + rv = request_irq(pdev->irq, ioh_i2s_irq, IRQF_SHARED, "ml7213_ioh", > >> + pdev); > >> + if (rv != 0) { > >> + printk(KERN_ERR "Failed to allocate irq\n"); > >> + goto out_irq; > >> + } > > Are you *sure* you're ready to handle interrupts at this point? > This is just registering interrupt handler. > As long as interrupt register is not enabled, the interrupt handler is > not called. > This is common request_irq description. > What's is your concern ? Apart from anything else you've got the interrupt requested as IRQF_SHARED so the interrupt could get called at any time. It's also not clear that you've got the hardware in a known good state.

3 28

[alsa-devel] [PATCH] alsa-lib/tlv: fix handling of raw value ranges
by Benoît Thébaudeau 02 Jun '12

02 Jun '12

In case of a TLV dB range with all items having raw value ranges strictly within the main raw value range reported by the driver, snd_tlv_convert_from_dB() returned one of the main raw range boundaries, which was outside all dB range items. E.g.: static const unsigned int bd3753x_vol_fader_gain_att_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), 48, 142, TLV_DB_SCALE_ITEM(-7900, 100, 0), }; SOC_SINGLE_TLV("Main Volume", BD3753X_VOL_GAIN, 0, 255, 1, bd3753x_vol_fader_gain_att_tlv), snd_tlv_convert_from_dB(tlv, 0, 255, 1500, &value, 1) returned 255 instead of the expected 142. or: static const unsigned int max9850_tlv[] = { TLV_DB_RANGE_HEAD(4), 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), 0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0), 0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0), 0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0), }; SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv), snd_tlv_convert_from_dB(tlv, 0, 0x3f, -7450, &value, 0) returned 0 instead of the expected 0x18. Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- alsa-lib/src/control/tlv.c +++ alsa-lib/src/control/tlv.c @@ -291,41 +291,37 @@ int snd_tlv_convert_from_dB(unsigned int { switch (tlv[0]) { case SND_CTL_TLVT_DB_RANGE: { - long dbmin, dbmax, prev_rangemax; + long dbmin, dbmax, prev_submax; unsigned int pos, len; len = int_index(tlv[1]); - if (len > MAX_TLV_RANGE_SIZE) - return -EINVAL; - if (snd_tlv_get_dB_range(tlv, rangemin, rangemax, - &dbmin, &dbmax)) + if (len < 6 || len > MAX_TLV_RANGE_SIZE) return -EINVAL; - if (db_gain <= dbmin) { - *value = rangemin; - return 0; - } else if (db_gain >= dbmax) { - *value = rangemax; - return 0; - } pos = 2; - prev_rangemax = 0; - while (pos + 4 <= len) { - rangemin = (int)tlv[pos]; - rangemax = (int)tlv[pos + 1]; + prev_submax = 0; + do { + long submin, submax; + submin = (int)tlv[pos]; + submax = (int)tlv[pos + 1]; + if (rangemax < submax) + submax = rangemax; if (!snd_tlv_get_dB_range(tlv + pos + 2, - rangemin, rangemax, + submin, submax, &dbmin, &dbmax) && db_gain >= dbmin && db_gain <= dbmax) return snd_tlv_convert_from_dB(tlv + pos + 2, - rangemin, rangemax, + submin, submax, db_gain, value, xdir); else if (db_gain < dbmin) { - *value = xdir ? rangemin : prev_rangemax; + *value = xdir || pos == 2 ? submin : prev_submax; return 0; } - prev_rangemax = rangemax; + prev_submax = submax; + if (rangemax == submax) + break; pos += int_index(tlv[pos + 3]) + 4; - } - return -EINVAL; + } while (pos + 4 <= len); + *value = prev_submax; + return 0; } case SND_CTL_TLVT_DB_SCALE: { int min, step, max;

4 24

[alsa-devel] ASoC: wm9712: Microphone doesn't work, "Capture Volume" inverted
by Christoph Fritz 15 May '12

15 May '12

Hi Mark, Liam, Hans J. and List, I have a "wm9712" codec-chip on an ARM mx35 board and everything works except Microphone. Strange is that I can't change the Mic-Select-Source: # amixer sset "Mic Select Source" 'Mic 2' Simple mixer control 'Mic Select Source',0 Capabilities: enum Items: 'Mic 1' 'Differential' 'Mic 2' 'Stereo' Item0: 'Mic 1' When I tested arecord for line-in, the "Capture Slide" in alsamixer was wrongly inverted. in sound/soc/codecs/wm9712.c: SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), should get: SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), If you want I can send a patch in. I'm using the current linus git tree (3.4.0-rc4). I tried it with linux-2.6.33.20 (with sound/soc/imx/imx-ssi.c etc. from 2.6.34) and it works. On a fairly similar board I was told that it works on Kernel 3.0.4 with this patch applied: Since patch "ASoC: Sync initial widget state with hardware" (commitid: b66a70d5e9929f3b0df5a7177bba75652d2f4c3e) the mic capture doesn't work. The MICBIAS bit from the "Power Management" Register are alway 1 (disabled). So that the captering works the active status must be set to 1, at the right time. That this works the stream name must be set. --- include/sound/soc-dapm.h | 3 ++- sound/soc/soc-dapm.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c46e7d8..308122a 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -85,7 +85,8 @@ .shift = wshift, .invert = winvert, .kcontrol_news = wcontrols, \ .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ -{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ +{ .id = snd_soc_dapm_micbias, .name = wname, .sname = \ + "Mic Bias HiFi Capture", .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrol_news = NULL, .num_kcontrols = 0} #define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \ { .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 32ab7fc..250a1d2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -646,7 +646,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return snd_soc_dapm_suspend_check(widget); /* connected VMID/Bias for lower pops */ - if (widget->id == snd_soc_dapm_vmid) + if (widget->id == snd_soc_dapm_vmid || widget->id == snd_soc_dapm_micbias) return snd_soc_dapm_suspend_check(widget); /* connected jack ? */ @@ -1835,6 +1835,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_micbias: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: @@ -1848,7 +1849,6 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) break; case snd_soc_dapm_input: case snd_soc_dapm_output: - case snd_soc_dapm_micbias: case snd_soc_dapm_spk: case snd_soc_dapm_hp: case snd_soc_dapm_mic: -- Thanks, -- Christoph It follows the output of "amixer scontents", "amixer controls" and "amixer contents": Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 20 [65%] [on] Front Right: Playback 20 [65%] [on] Simple mixer control 'Headphone Playback ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 6 [19%] [off] Front Right: Playback 6 [19%] [off] Simple mixer control 'Speaker Mixer Aux',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Speaker Mixer Line Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Speaker Mixer PCBeep Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Speaker Mixer PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Speaker Mixer Phone Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Speaker Mux',0 Capabilities: enum Items: 'Speaker Mix' 'Headphone Mix' Item0: 'Speaker Mix' Simple mixer control 'Speaker Playback Invert',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Speaker Playback ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Tone Cut-off',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Bass',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control 'Bass Control',0 Capabilities: enum Items: 'Linear Control' 'Adaptive Boost' Item0: 'Linear Control' Simple mixer control 'Bass Cut-off',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Treble',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 9 [60%] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 17 [55%] Front Right: Playback 17 [55%] Simple mixer control 'Line',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] Front Right: Capture 23 [74%] Simple mixer control 'Mic 1',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 31 Mono: 23 [74%] [0.00dB] Simple mixer control 'Mic 2',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 31 Mono: 18 [58%] [-7.50dB] Simple mixer control 'Mic Boost',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 1 Mono: 0 [0%] [0.00dB] Simple mixer control 'Mic Headphone ',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 3 [43%] Simple mixer control 'Mic Select Source',0 Capabilities: enum Items: 'Mic 1' 'Differential' 'Mic 2' 'Stereo' Item0: 'Mic 1' Simple mixer control 'Phone',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 14 [93%] Simple mixer control 'Phone Mixer Aux',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Phone Mixer Line Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Phone Mixer Mic 1 Sidetone',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Phone Mixer Mic 2 Sidetone',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Phone Mixer PCBeep Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Phone Mixer PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Aux Playback Headphone',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'Aux Playback Phone',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'Aux Playback Speaker',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 0 [0%] [off] Simple mixer control 'Mono Playback ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Playback Attenuate (-6dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Capture',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 63 Front Left: Capture 13 [21%] Front Right: Capture 13 [21%] Simple mixer control 'Capture 20dB Boost',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Capture ADC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture Phone Mux',0 Capabilities: enum Items: 'Stereo' 'Left' 'Right' 'Mute' Item0: 'Mute' Simple mixer control 'Capture Volume Steps',0 Capabilities: enum Items: '+1.5dB Steps' '+0.75dB Steps' Item0: '+0.75dB Steps' Item1: '+0.75dB Steps' Simple mixer control 'Capture ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture to Phone 20dB Boost',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control '3D',0 Capabilities: pvolume pvolume-joined penum Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 0 [0%] Simple mixer control '3D Lower Cut-off',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control '3D Upper Cut-off',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'ALC Attack Time',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control 'ALC Decay Time',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control 'ALC Function',0 Capabilities: enum Items: 'None' 'Left' 'Right' 'Stereo' Item0: 'None' Simple mixer control 'ALC Headphone',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'ALC Hold Time',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control 'ALC Max',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'ALC NG',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'ALC NG Threshold',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 31 Mono: 0 [0%] Simple mixer control 'ALC NG Type',0 Capabilities: enum Items: 'Constant Gain' 'Mute' Item0: 'Mute' Simple mixer control 'ALC Sidetone Mux',0 Capabilities: enum Items: 'Stereo' 'Left' 'Right' 'None' Item0: 'Stereo' Simple mixer control 'ALC Target',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control 'ALC ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'ALC ZC Timeout',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 0 [0%] Simple mixer control 'Differential Source',0 Capabilities: enum Items: 'Mic' 'Line' Item0: 'Mic' Simple mixer control 'Left Capture Select',0 Capabilities: enum Items: 'Mic' 'NC' 'NC' 'Speaker Mixer' 'Line' 'Headphone Mixer' 'Phone Mixer' 'Phone' Item0: 'NC' Simple mixer control 'Left HP Mixer Aux',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Left HP Mixer Line Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Left HP Mixer Mic Sidetone',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Left HP Mixer PCBeep Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Left HP Mixer PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Left HP Mixer Phone Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Out3',0 Capabilities: volume volume-joined pswitch pswitch-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 31 Mono: 0 [0%] Playback [off] Simple mixer control 'Out3 Mux',0 Capabilities: enum Items: 'Left' 'VREF' 'Left + Right' 'Mono' Item0: 'Mono' Simple mixer control 'Out3 ZC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PCBeep Bypass Headphone',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'PCBeep Bypass Phone',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'PCBeep Bypass Speaker',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 7 Mono: 0 [0%] Simple mixer control 'Right Capture Select',0 Capabilities: enum Items: 'Mic' 'NC' 'NC' 'Speaker Mixer' 'Line' 'Headphone Mixer' 'Phone Mixer' 'Phone' Item0: 'NC' Simple mixer control 'Right HP Mixer Aux',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Right HP Mixer Line Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Right HP Mixer Mic Sidetone',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Right HP Mixer PCBeep Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Right HP Mixer PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Right HP Mixer Phone Bypass',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] numid=8,iface=MIXER,name='Headphone Playback ZC Switch' numid=4,iface=MIXER,name='Headphone Playback Switch' numid=3,iface=MIXER,name='Headphone Playback Volume' numid=5,iface=MIXER,name='PCM Playback Volume' numid=35,iface=MIXER,name='Line Capture Volume' numid=51,iface=MIXER,name='Mic 1 Volume' numid=52,iface=MIXER,name='Mic 2 Volume' numid=53,iface=MIXER,name='Mic Boost Volume' numid=23,iface=MIXER,name='Mic Headphone Volume' numid=78,iface=MIXER,name='Mic Select Source' numid=60,iface=MIXER,name='Phone Mixer Aux Playback Switch' numid=61,iface=MIXER,name='Phone Mixer Line Bypass Switch' numid=63,iface=MIXER,name='Phone Mixer Mic 1 Sidetone Switch' numid=64,iface=MIXER,name='Phone Mixer Mic 2 Sidetone Switch' numid=59,iface=MIXER,name='Phone Mixer PCBeep Bypass Switch' numid=62,iface=MIXER,name='Phone Mixer PCM Playback Switch' numid=34,iface=MIXER,name='Phone Volume' numid=31,iface=MIXER,name='Aux Playback Headphone Volume' numid=33,iface=MIXER,name='Aux Playback Phone Volume' numid=32,iface=MIXER,name='Aux Playback Speaker Volume' numid=9,iface=MIXER,name='Mono Playback ZC Switch' numid=11,iface=MIXER,name='Mono Playback Switch' numid=10,iface=MIXER,name='Mono Playback Volume' numid=36,iface=MIXER,name='Capture 20dB Boost Switch' numid=47,iface=MIXER,name='Capture ADC Switch' numid=81,iface=MIXER,name='Capture Phone Mux' numid=50,iface=MIXER,name='Capture ZC Switch' numid=37,iface=MIXER,name='Capture to Phone 20dB Boost Switch' numid=49,iface=MIXER,name='Capture Volume' numid=48,iface=MIXER,name='Capture Volume Steps' numid=44,iface=MIXER,name='Playback Attenuate (-6dB) Switch' numid=39,iface=MIXER,name='3D Lower Cut-off Switch' numid=40,iface=MIXER,name='3D Playback Volume' numid=38,iface=MIXER,name='3D Upper Cut-off Switch' numid=15,iface=MIXER,name='ALC Attack Time' numid=14,iface=MIXER,name='ALC Decay Time' numid=16,iface=MIXER,name='ALC Function' numid=24,iface=MIXER,name='ALC Headphone Volume' numid=13,iface=MIXER,name='ALC Hold Time' numid=17,iface=MIXER,name='ALC Max Volume' numid=20,iface=MIXER,name='ALC NG Switch' numid=22,iface=MIXER,name='ALC NG Threshold' numid=21,iface=MIXER,name='ALC NG Type' numid=84,iface=MIXER,name='ALC Sidetone Mux' numid=12,iface=MIXER,name='ALC Target Volume' numid=19,iface=MIXER,name='ALC ZC Switch' numid=18,iface=MIXER,name='ALC ZC Timeout' numid=41,iface=MIXER,name='Bass Control' numid=42,iface=MIXER,name='Bass Cut-off Switch' numid=45,iface=MIXER,name='Bass Volume' numid=77,iface=MIXER,name='Differential Source' numid=80,iface=MIXER,name='Left Capture Select' numid=72,iface=MIXER,name='Left HP Mixer Aux Playback Switch' numid=74,iface=MIXER,name='Left HP Mixer Line Bypass Switch' numid=76,iface=MIXER,name='Left HP Mixer Mic Sidetone Switch' numid=71,iface=MIXER,name='Left HP Mixer PCBeep Bypass Switch' numid=75,iface=MIXER,name='Left HP Mixer PCM Playback Switch' numid=73,iface=MIXER,name='Left HP Mixer Phone Bypass Switch' numid=83,iface=MIXER,name='Out3 Mux' numid=25,iface=MIXER,name='Out3 Switch' numid=27,iface=MIXER,name='Out3 Volume' numid=26,iface=MIXER,name='Out3 ZC Switch' numid=28,iface=MIXER,name='PCBeep Bypass Headphone Volume' numid=30,iface=MIXER,name='PCBeep Bypass Phone Volume' numid=29,iface=MIXER,name='PCBeep Bypass Speaker Volume' numid=79,iface=MIXER,name='Right Capture Select' numid=66,iface=MIXER,name='Right HP Mixer Aux Playback Switch' numid=68,iface=MIXER,name='Right HP Mixer Line Bypass Switch' numid=70,iface=MIXER,name='Right HP Mixer Mic Sidetone Switch' numid=65,iface=MIXER,name='Right HP Mixer PCBeep Bypass Switch' numid=69,iface=MIXER,name='Right HP Mixer PCM Playback Switch' numid=67,iface=MIXER,name='Right HP Mixer Phone Bypass Switch' numid=55,iface=MIXER,name='Speaker Mixer Aux Playback Switch' numid=57,iface=MIXER,name='Speaker Mixer Line Bypass Switch' numid=54,iface=MIXER,name='Speaker Mixer PCBeep Bypass Switch' numid=58,iface=MIXER,name='Speaker Mixer PCM Playback Switch' numid=56,iface=MIXER,name='Speaker Mixer Phone Bypass Switch' numid=82,iface=MIXER,name='Speaker Mux' numid=7,iface=MIXER,name='Speaker Playback Invert Switch' numid=2,iface=MIXER,name='Speaker Playback Switch' numid=1,iface=MIXER,name='Speaker Playback Volume' numid=6,iface=MIXER,name='Speaker Playback ZC Switch' numid=43,iface=MIXER,name='Tone Cut-off Switch' numid=46,iface=MIXER,name='Treble Volume' numid=8,iface=MIXER,name='Headphone Playback ZC Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=4,iface=MIXER,name='Headphone Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='Headphone Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=31,step=0 : values=20,20 numid=5,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=31,step=0 : values=17,17 numid=35,iface=MIXER,name='Line Capture Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=31,step=0 : values=23,23 numid=51,iface=MIXER,name='Mic 1 Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=31,step=0 : values=23 | dBscale-min=-34.50dB,step=1.50dB,mute=0 numid=52,iface=MIXER,name='Mic 2 Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=31,step=0 : values=18 | dBscale-min=-34.50dB,step=1.50dB,mute=0 numid=53,iface=MIXER,name='Mic Boost Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=1,step=0 : values=0 | dBscale-min=0.00dB,step=20.00dB,mute=0 numid=23,iface=MIXER,name='Mic Headphone Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=3 numid=78,iface=MIXER,name='Mic Select Source' ; type=ENUMERATED,access=rw------,values=1,items=4 ; Item #0 'Mic 1' ; Item #1 'Differential' ; Item #2 'Mic 2' ; Item #3 'Stereo' : values=0 numid=60,iface=MIXER,name='Phone Mixer Aux Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=61,iface=MIXER,name='Phone Mixer Line Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=63,iface=MIXER,name='Phone Mixer Mic 1 Sidetone Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=64,iface=MIXER,name='Phone Mixer Mic 2 Sidetone Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=59,iface=MIXER,name='Phone Mixer PCBeep Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=62,iface=MIXER,name='Phone Mixer PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=34,iface=MIXER,name='Phone Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=14 numid=31,iface=MIXER,name='Aux Playback Headphone Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=33,iface=MIXER,name='Aux Playback Phone Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=32,iface=MIXER,name='Aux Playback Speaker Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=9,iface=MIXER,name='Mono Playback ZC Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=11,iface=MIXER,name='Mono Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Mono Playback Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=31,step=0 : values=0 numid=36,iface=MIXER,name='Capture 20dB Boost Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=47,iface=MIXER,name='Capture ADC Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=81,iface=MIXER,name='Capture Phone Mux' ; type=ENUMERATED,access=rw------,values=1,items=4 ; Item #0 'Stereo' ; Item #1 'Left' ; Item #2 'Right' ; Item #3 'Mute' : values=3 numid=50,iface=MIXER,name='Capture ZC Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=37,iface=MIXER,name='Capture to Phone 20dB Boost Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=49,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=63,step=0 : values=13,13 numid=48,iface=MIXER,name='Capture Volume Steps' ; type=ENUMERATED,access=rw------,values=2,items=2 ; Item #0 '+1.5dB Steps' ; Item #1 '+0.75dB Steps' : values=1,1 numid=44,iface=MIXER,name='Playback Attenuate (-6dB) Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=39,iface=MIXER,name='3D Lower Cut-off Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=40,iface=MIXER,name='3D Playback Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=0 numid=38,iface=MIXER,name='3D Upper Cut-off Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=15,iface=MIXER,name='ALC Attack Time' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=0 numid=14,iface=MIXER,name='ALC Decay Time' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=0 numid=16,iface=MIXER,name='ALC Function' ; type=ENUMERATED,access=rw------,values=1,items=4 ; Item #0 'None' ; Item #1 'Left' ; Item #2 'Right' ; Item #3 'Stereo' : values=0 numid=24,iface=MIXER,name='ALC Headphone Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=13,iface=MIXER,name='ALC Hold Time' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=0 numid=17,iface=MIXER,name='ALC Max Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=20,iface=MIXER,name='ALC NG Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=22,iface=MIXER,name='ALC NG Threshold' ; type=INTEGER,access=rw------,values=1,min=0,max=31,step=0 : values=0 numid=21,iface=MIXER,name='ALC NG Type' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Constant Gain' ; Item #1 'Mute' : values=1 numid=84,iface=MIXER,name='ALC Sidetone Mux' ; type=ENUMERATED,access=rw------,values=1,items=4 ; Item #0 'Stereo' ; Item #1 'Left' ; Item #2 'Right' ; Item #3 'None' : values=0 numid=12,iface=MIXER,name='ALC Target Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=0 numid=19,iface=MIXER,name='ALC ZC Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=18,iface=MIXER,name='ALC ZC Timeout' ; type=INTEGER,access=rw------,values=1,min=0,max=3,step=0 : values=0 numid=41,iface=MIXER,name='Bass Control' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Linear Control' ; Item #1 'Adaptive Boost' : values=0 numid=42,iface=MIXER,name='Bass Cut-off Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=45,iface=MIXER,name='Bass Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=0 numid=77,iface=MIXER,name='Differential Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Mic' ; Item #1 'Line' : values=0 numid=80,iface=MIXER,name='Left Capture Select' ; type=ENUMERATED,access=rw------,values=1,items=8 ; Item #0 'Mic' ; Item #1 'NC' ; Item #2 'NC' ; Item #3 'Speaker Mixer' ; Item #4 'Line' ; Item #5 'Headphone Mixer' ; Item #6 'Phone Mixer' ; Item #7 'Phone' : values=1 numid=72,iface=MIXER,name='Left HP Mixer Aux Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=74,iface=MIXER,name='Left HP Mixer Line Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=76,iface=MIXER,name='Left HP Mixer Mic Sidetone Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=71,iface=MIXER,name='Left HP Mixer PCBeep Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=75,iface=MIXER,name='Left HP Mixer PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=73,iface=MIXER,name='Left HP Mixer Phone Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=83,iface=MIXER,name='Out3 Mux' ; type=ENUMERATED,access=rw------,values=1,items=4 ; Item #0 'Left' ; Item #1 'VREF' ; Item #2 'Left + Right' ; Item #3 'Mono' : values=3 numid=25,iface=MIXER,name='Out3 Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=27,iface=MIXER,name='Out3 Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=31,step=0 : values=0 numid=26,iface=MIXER,name='Out3 ZC Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=28,iface=MIXER,name='PCBeep Bypass Headphone Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=30,iface=MIXER,name='PCBeep Bypass Phone Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=29,iface=MIXER,name='PCBeep Bypass Speaker Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=7,step=0 : values=0 numid=79,iface=MIXER,name='Right Capture Select' ; type=ENUMERATED,access=rw------,values=1,items=8 ; Item #0 'Mic' ; Item #1 'NC' ; Item #2 'NC' ; Item #3 'Speaker Mixer' ; Item #4 'Line' ; Item #5 'Headphone Mixer' ; Item #6 'Phone Mixer' ; Item #7 'Phone' : values=1 numid=66,iface=MIXER,name='Right HP Mixer Aux Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=68,iface=MIXER,name='Right HP Mixer Line Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=70,iface=MIXER,name='Right HP Mixer Mic Sidetone Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=65,iface=MIXER,name='Right HP Mixer PCBeep Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=69,iface=MIXER,name='Right HP Mixer PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=67,iface=MIXER,name='Right HP Mixer Phone Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=55,iface=MIXER,name='Speaker Mixer Aux Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=57,iface=MIXER,name='Speaker Mixer Line Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=54,iface=MIXER,name='Speaker Mixer PCBeep Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=58,iface=MIXER,name='Speaker Mixer PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=56,iface=MIXER,name='Speaker Mixer Phone Bypass Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=82,iface=MIXER,name='Speaker Mux' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Speaker Mix' ; Item #1 'Headphone Mix' : values=0 numid=7,iface=MIXER,name='Speaker Playback Invert Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=2,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=1,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=31,step=0 : values=6,6 numid=6,iface=MIXER,name='Speaker Playback ZC Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=43,iface=MIXER,name='Tone Cut-off Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=46,iface=MIXER,name='Treble Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=15,step=0 : values=9

2 16

[alsa-devel] Distorted jackd output on Macbooks
by Adrian Knoth 12 May '12

12 May '12

Hi! JFTR, we've seen users that report distorted sound on their macbooks when using jackd. The workaround is to force 16-bit output: http://jackaudio.org/macbook_distortion Chip in question was the Cirrus Logic CS4206. We don't know yet which of the formats (32/24) is broken, maybe only one, maybe both. I happen to have a spare MacBookPro around, but I won't have time to look into that myself anytime soon. If somebody else has such a device and wants to give it a whirl, feel free to go ahead. ;) Cheers

2 5

[alsa-devel] [PATCH 0/3] ARM: OMAP: Add platform devices for ASoC HDMI drivers
by Ricardo Neri 09 May '12

09 May '12

This set of patches is intended to add the platform devices for the ASoC HDMI drivers when not using device tree. For this, I took an approach similar to DMIC and McPDM by creating a dedicated functions to create the devices. I included the device for the CPU DAI driver, omap-hdmi-audio-dai, and the device for the machine driver, omap-hdmi-audio, in devices.c to reflect the fact that any OMAP4 (and OMAP5 in the future) has HDMI IP. I put the device for the codec in the board file to reflect the fact that not necessarily all the boards with OMAP4/5 have the HDMI output wired. Best regards Ricardo Ricardo Neri (3): ARM: OMAP: devices: Register platform devices for HDMI audio ARM: OMAP4: board-4430sdp: Register platform device for HDMI audio codec ARM: OMAP4: board-omap4panda: Register platform device for HDMI audio codec arch/arm/mach-omap2/board-4430sdp.c | 6 ++++++ arch/arm/mach-omap2/board-omap4panda.c | 6 ++++++ arch/arm/mach-omap2/devices.c | 31 +++++++++++++++++++++++++++++++ 3 files changed, 43 insertions(+), 0 deletions(-) -- 1.7.5.4

2 6

[alsa-devel] Issue with RME HDSPe AIO latency
by jamesm 07 May '12

07 May '12

Hello all, I'm using the RME hdspm kernel module with a Hammerfall HDSPe AIO card. I'm on RHEL 5.5, with alsa 1.0.17, kernel 2.6.18-194.11.4. I have backported the latest alsa hdspm code to this kernel, as there were a number of bugs in the one that ships with the 2.6.18-194 kernel. I've also implemented support for copy() and silence() functions so that the hdspm driver can be used with applications via the RDWR interface instead of just MMAP (ie it just works with aplay without using plughw first.) The latest hdspm driver code from alsa and kernel.org indicates that the HDSPe AIO card has a fixed buffer size of 16384. I've chosen a period size of 1024, which gives me 16 periods. The issue I have is that whenever I play audio through the hdspm driver, the delay between the hw pointer and sw pointer is always close to the maximum buffer size (approx 16000). This leaves me with a large delay between writing data into the PCM and it coming out of the card. I'd like this to be between 1024 and 2048, I could probably crank the period down further to 512. My application sets the start_threshold to 0 and avail_min to one period (1024). It then runs a thread, constantly writing data to the PCM (silence or real data). Does anyone have any suggestions as to why the hdspm driver hw pointer is not following the application pointer more closely? Is it due to the number of periods? Unfortunately I can't change that as the AIO buffer size is fixed. Here are my pcm hwparams, swparams and status from /proc/asound when my app is running: $ cat /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: RME AIO name: RME AIO subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 0 $ cat /proc/asound/card0/pcm0p/sub0/hw_params access: RW_NONINTERLEAVED format: S32_LE subformat: STD channels: 16 rate: 48000 (48000/1) period_size: 1024 buffer_size: 16384 tick_time: 1000 $ cat /proc/asound/card0/pcm0p/sub0/sw_params tstamp_mode: NONE period_step: 1 sleep_min: 0 avail_min: 1024 xfer_align: 1 start_threshold: 0 stop_threshold: 18446744073709551615 silence_threshold: 0 silence_size: 0 boundary: 4611686018427387904 $ cat /proc/asound/card0/pcm0p/sub0/status state: RUNNING trigger_time: 1335799965.972782000 tstamp : 1335801526.687332000 delay : 15664 avail : 720 avail_max : 1056 ----- hw_ptr : 74911440 appl_ptr : 74927104 Regards James Milne FilmLight Ltd.

3 2

[alsa-devel] [PATCH] pci/echoaudio: Remove incorrect part of assertion
by Mark Hills 06 May '12

06 May '12

This assertion seems to imply that chip->dsp_code_to_load is a pointer. It's actually an integer handle on the actual firmware, and 0 has no special meaning. The assertion prevents initialisation of a Darla20 card, but would also affect other models. It seems it was introduced in commit dd7b254d. ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting... ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000 ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware... ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20 ------------[ cut here ]------------ WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]() Hardware name: Dell DM051 BUG? (!chip->dsp_code_to_load || !chip->comm_page) Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/pci/echoaudio/echoaudio_dsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 64417a7..d8c670c 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip) const struct firmware *fw; int box_type, err; - if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page)) + if (snd_BUG_ON(!chip->comm_page)) return -EPERM; /* See if the ASIC is present and working - only if the DSP is already loaded */ -- 1.7.4.4

2 1

[alsa-devel] Call for propsals: Audio micro-conf at LPC 2012 San Diego
by Takashi Iwai 05 May '12

05 May '12

Hi, we're planning to hold an Audio micro-conference again this year at Linux Plumbers conference (LPC). It'll be held at San Diego, CA, Aug. 29-31. The Wiki page is found at http://wiki.linuxplumbersconf.org/2012:audio The items listed there are rough ideas popping up during a mail thread. If you have interest and want to give a talk or discussions, don't hesitate to submit your proposal. Please mail to submissions(a)linuxplumbersconf.org, as described in the page below: http://www.linuxplumbersconf.org/2012/2012-lpc-call-for-proposals-take-2/ The deadline for submission is 11:59 PDT on May 1, 2012. So, it's pretty little time left. Sorry for the late announce... thanks, Takashi

2 2

[alsa-devel] [PATCH 00/11] enable imx6q_sabrelite sgtl5000 audio support
by Richard Zhao 02 May '12

02 May '12

It's based on Dong Aisheng's early pinctrl patch and Sascha's v2 convert to common clk patch. You can also get at: https://github.com/riczhao/kernel-imx/tree/topics/audio Richard Zhao (11): dma: imx-sdma: make channel0 operations atomic ASoC: imx-sgtl5000: add of_node_put when probe fail. ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000 i2c: imx: add pinctrl support ARM: imx6q: move imx6q_sabrelite specific code to a dedicated function ARM: dts: imx6q-sabrelite: add ssi device ARM: dts: imx6q-sabrelite: add audmux device ASoC: imx-audmux: add pinctrl support ARM: imx6q: add ssi1 clk_lookup ARM: imx6q_sabrelite: clkdev_add cko1 for sgtl5000 ARM: dts: imx6q-sabrelite: add sound device imx6q-sabrelite-sgtl5000 arch/arm/boot/dts/imx6q-sabrelite.dts | 30 ++++++++++++++++ arch/arm/boot/dts/imx6q.dtsi | 46 +++++++++++++++++++++++-- arch/arm/mach-imx/clk-imx6q.c | 4 ++ arch/arm/mach-imx/mach-imx6q.c | 37 +++++++++++++++++++- drivers/dma/imx-sdma.c | 57 ++++++++++++++++++------------- drivers/i2c/busses/i2c-imx.c | 9 +++++ sound/soc/fsl/imx-audmux.c | 8 ++++ sound/soc/fsl/imx-sgtl5000.c | 61 +++++++++++++++++++++++++-------- 8 files changed, 208 insertions(+), 44 deletions(-) Thanks Richard -- 1.7.5.4

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