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[alsa-devel] [PATCH 1/2] ASoC: Refector DAPM event handler
by Mark Brown 17 Jul '08

17 Jul '08

The DAPM event callback code has many layers of indentation, taking it over 80 columns. Refactor the code to give less indentation in order to avoid checkpatch issues on further changes and exploding indentation. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 79 ++++++++++++++++++++++++++------------------------ 1 files changed, 41 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061..17698ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -586,45 +586,48 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) power_change = (w->power == power) ? 0: 1; w->power = power; + if (!power_change) + continue; + /* call any power change event handlers */ - if (power_change) { - if (w->event) { - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); - if (power) { - /* power up event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - } else { - /* power down event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - } - } else - /* no event handler */ - dapm_update_bits(w); + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + dapm_update_bits(w); + + /* power up post event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; } } } -- 1.5.6.2

1 1

[alsa-devel] snd_pcm_writei() -last value sustained
by Alan Horstmann 17 Jul '08

17 Jul '08

Working on an app I am seeing several strange effects, and this is one of them. It can be demonstrated with the pcm.c example program, and I have attempted to attach a small shot of Audacity capturing the effect. After writing to the pcm with snd_pcm_writei() a certain amount of data (I have pcm.c doing 2 periods) the value captured from the soundcard remains at the last value thereafter; clearly this card doesn't have DC elimination or this would not be seen. The sound card is ice1712 (DMX6fire); I had always noticed the PCM meters in envy24control often stayed high if something like XMMS was stopped or paused during play, but thought it a metering fault. Now it appears to be actual pcm value that 'gets stuck'. Close examination with Audacity shows it is in fact the last-but-one value that is sustained. Is this expected behaviour, configuration fault, driver bug or other problem? I had expected that after the block of data had been written out as audio the signal would return to zero straightaway. Can anyone give me any clues? Alan

2 6

[alsa-devel] [PATCH 1/2] ASoC: Refector DAPM event handler
by Mark Brown 17 Jul '08

17 Jul '08

The DAPM event callback code has many layers of indentation. Refector the code to give less indentation in order to facilitiate further refactoring. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 76 +++++++++++++++++++++++++------------------------- 1 files changed, 38 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061..9307b10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -587,44 +587,44 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) w->power = power; /* call any power change event handlers */ - if (power_change) { - if (w->event) { - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); - if (power) { - /* power up event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - } else { - /* power down event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - } - } else - /* no event handler */ - dapm_update_bits(w); + if (power_change && w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power_change && power && w->event && + w->event_flags & SND_SOC_DAPM_PRE_PMU) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (power_change && !power && w->event && + w->event_flags & SND_SOC_DAPM_PRE_PMD) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + dapm_update_bits(w); + + /* power up post event */ + if (power_change && power && w->event && + w->event_flags & SND_SOC_DAPM_POST_PMU){ + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (power_change && !power && w->event && + w->event_flags & SND_SOC_DAPM_POST_PMD) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; } } } -- 1.5.6.2

2 7

[alsa-devel] [PATCH] ALSA: ac97: copy subsys values from the PCI device
by Rene Herman 17 Jul '08

17 Jul '08

Hi. Just getting to know to the code... Why do we reread the subsys values from PCI config space in the AC97 code? (this in addition to why we store the values in the ac97 struct in the first place if they're available through ac97->pci but that's secondary) Rene.

2 2

[PATCH] ALSA: ac97: copy subsys values from the PCI device
by Rene Herman 17 Jul '08

17 Jul '08

copy the subsys values from the PCI device instead of rereading them. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ac97/ac97_codec.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 45fd290..4f4343d 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2015,8 +2015,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, #ifdef CONFIG_PCI if (ac97->pci) { - pci_read_config_word(ac97->pci, PCI_SUBSYSTEM_VENDOR_ID, &ac97->subsystem_vendor); - pci_read_config_word(ac97->pci, PCI_SUBSYSTEM_ID, &ac97->subsystem_device); + ac97->subsystem_vendor = ac97->pci->subsystem_vendor; + ac97->subsystem_device = ac97->pci->subsystem_device; } #endif if (bus->ops->reset) { -- 1.5.5 --------------000002040805080104050207--

1 0

[alsa-devel] [PATCH] ALSA: ens1370: communicate PCI device to AC97
by Rene Herman 17 Jul '08

17 Jul '08

Currently the subsys_vendor/device values always display as 0x0000 in the codec97#n/ac97#m-k file at least.

1 0

[PATCH] ALSA: ens1370: communicate PCI device to AC97
by Rene Herman 17 Jul '08

17 Jul '08

communicate the ES137x PCI device to the AC97 code for its subsys_vendor/device values Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ens1370.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 38808fb..7298fe0 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1629,6 +1629,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq, memset(&ac97, 0, sizeof(ac97)); ac97.private_data = ensoniq; ac97.private_free = snd_ensoniq_mixer_free_ac97; + ac97.pci = ensoniq->pci; ac97.scaps = AC97_SCAP_AUDIO; if ((err = snd_ac97_mixer(pbus, &ac97, &ensoniq->u.es1371.ac97)) < 0) return err; -- 1.5.5 --------------070204000106090400030805--

1 0

[alsa-devel] PCM/ALSA loopback (codec-->DMA_RX-->DMA_TX-->codec)
by Norbert van Bolhuis 17 Jul '08

17 Jul '08

to be able to easily check our entire analog audio in/out cicuits (with a sweep generator feeding mic in and measuring the signal from audio out) I'd like to know whether there's any low level loopback available in ALSA core (or ALSA lib). The closer to HW the better it is. This is what our analog/digital audio path look like: mic in -> analog in circuitry -> CODEC -> analog out circuitry -> out | | v ^ | | ----- CPU ----- kernel alsa core --------- alsa lib aplay/arecord The problem is there's no loopback/echo nor sidetone possible with the CODEC we use. Also the DMA between CPU and CODEC cannot be looped (no DMA_RX-->DMA_TX possible). I know it's possible to loop digital audio in user-space with arecord -D hw:0,0 .. | aplay -D hw:0,0 .. This will probably give latencies of 10-100ms (depending upon CPU, DMA priority, SW load, period_size, etc...). These are too high for the sweep generator. Is it possible to loopback a PCM in stream to a PCM out stream somehow in ALSA core/lib ? How does the latency ALSA lib example program do this ? -- This message has been scanned for viruses and is believed to be clean

1 0

[alsa-devel] [PATCH 5/xx] wss_lib: use wss constants instead of ad1848 ones
by Krzysztof Helt 17 Jul '08

17 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use wss constants for mode. Move ad1848 hardware constants to the wss.h. Move mixer tlv macros into the ad1848_lib.c from the ad1848.h. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Another step toward unification of the ad1848_lib and wss_lib. I would like to ask Rene for gathering all these patches into his git tree so they can be pushed from there once the unification is finished. This is the fifth patch. The previous four are here: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008766.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008669.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008978.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008901.html diff -urp linux-alsa/include/sound/ad1848.h linux-2.6.25/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/include/sound/ad1848.h 2008-07-08 21:43:51.000000000 +0200 @@ -104,24 +104,6 @@ #define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ #define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ -/* defines for codec.mode */ - -#define AD1848_MODE_NONE 0x0000 -#define AD1848_MODE_PLAY 0x0001 -#define AD1848_MODE_CAPTURE 0x0002 -#define AD1848_MODE_TIMER 0x0004 -#define AD1848_MODE_OPEN (AD1848_MODE_PLAY|AD1848_MODE_CAPTURE|AD1848_MODE_TIMER) -#define AD1848_MODE_RUNNING 0x0010 - -/* defines for codec.hardware */ - -#define AD1848_HW_DETECT 0x0000 /* let AD1848 driver detect chip */ -#define AD1848_HW_AD1847 0x0001 /* AD1847 chip */ -#define AD1848_HW_AD1848 0x0002 /* AD1848 chip */ -#define AD1848_HW_CS4248 0x0003 /* CS4248 chip */ -#define AD1848_HW_CMI8330 0x0004 /* CMI8330 chip */ -#define AD1848_HW_THINKPAD 0x0005 /* Thinkpad 360/750/755 */ - /* IBM Thinkpad specific stuff */ #define AD1848_THINKPAD_CTL_PORT1 0x15e8 #define AD1848_THINKPAD_CTL_PORT2 0x15e9 @@ -167,26 +149,12 @@ struct ad1848_mix_elem { .type = AD1848_MIX_SINGLE, \ .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert) } -#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_SINGLE, \ - .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ - .tlv = xtlv } - #define AD1848_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ { .name = xname, \ .index = xindex, \ .type = AD1848_MIX_DOUBLE, \ .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert) } -#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_DOUBLE, \ - .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ - .tlv = xtlv } - int snd_ad1848_add_ctl_elem(struct snd_wss *chip, const struct ad1848_mix_elem *c); diff -urp linux-alsa/include/sound/wss.h linux-2.6.25/include/sound/wss.h --- linux-alsa/include/sound/wss.h 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/include/sound/wss.h 2008-07-08 21:52:32.000000000 +0200 @@ -55,6 +55,12 @@ #define WSS_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ #define WSS_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ #define WSS_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ +#define WSS_HW_AD1848_MASK 0x0800 /* AD1848 serie (half duplex) */ +#define WSS_HW_AD1847 0x0801 /* AD1847 chip */ +#define WSS_HW_AD1848 0x0802 /* AD1848 chip */ +#define WSS_HW_CS4248 0x0803 /* CS4248 chip */ +#define WSS_HW_CMI8330 0x0804 /* CMI8330 chip */ +#define WSS_HW_THINKPAD 0x0805 /* Thinkpad 360/750/755 */ /* compatible, but clones */ #define WSS_HW_INTERWAVE 0x1000 /* InterWave chip */ #define WSS_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ diff -urp linux-alsa/sound/isa/ad1848/ad1848.c linux-2.6.25/sound/isa/ad1848/ad1848.c --- linux-alsa/sound/isa/ad1848/ad1848.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848.c 2008-07-08 18:06:22.000000000 +0200 @@ -96,7 +96,7 @@ static int __devinit snd_ad1848_probe(st return -EINVAL; error = snd_ad1848_create(card, port[n], irq[n], dma1[n], - thinkpad[n] ? AD1848_HW_THINKPAD : AD1848_HW_DETECT, &chip); + thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, &chip); if (error < 0) goto out; diff -urp linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-2.6.25/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848_lib.c 2008-07-08 21:43:51.000000000 +0200 @@ -283,14 +283,12 @@ static int snd_ad1848_trigger(struct snd return 0; } snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] |= what); - chip->mode |= AD1848_MODE_RUNNING; } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { if (!(chip->image[AD1848_IFACE_CTRL] & what)) { spin_unlock(&chip->reg_lock); return 0; } snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] &= ~what); - chip->mode &= ~AD1848_MODE_RUNNING; } else { result = -EINVAL; } @@ -378,7 +376,7 @@ static int snd_ad1848_open(struct snd_ws { unsigned long flags; - if (chip->mode & AD1848_MODE_OPEN) + if (chip->mode & WSS_MODE_OPEN) return -EAGAIN; snd_ad1848_mce_down(chip); @@ -570,11 +568,9 @@ static irqreturn_t snd_ad1848_interrupt( { struct snd_wss *chip = dev_id; - if ((chip->mode & AD1848_MODE_PLAY) && chip->playback_substream && - (chip->mode & AD1848_MODE_RUNNING)) + if ((chip->mode & WSS_MODE_PLAY) && chip->playback_substream) snd_pcm_period_elapsed(chip->playback_substream); - if ((chip->mode & AD1848_MODE_CAPTURE) && chip->capture_substream && - (chip->mode & AD1848_MODE_RUNNING)) + if ((chip->mode & WSS_MODE_RECORD) && chip->capture_substream) snd_pcm_period_elapsed(chip->capture_substream); outb(0, AD1848P(chip, STATUS)); /* clear global interrupt bit */ return IRQ_HANDLED; @@ -690,19 +686,19 @@ static int snd_ad1848_probe(struct snd_w } if (id != 1) return -ENODEV; /* no valid device found */ - if (chip->hardware == AD1848_HW_DETECT) { + if (chip->hardware == WSS_HW_DETECT) { if (ad1847) { - chip->hardware = AD1848_HW_AD1847; + chip->hardware = WSS_HW_AD1847; } else { - chip->hardware = AD1848_HW_AD1848; + chip->hardware = WSS_HW_AD1848; rev = snd_ad1848_in(chip, AD1848_MISC_INFO); if (rev & 0x80) { - chip->hardware = AD1848_HW_CS4248; + chip->hardware = WSS_HW_CS4248; } else if ((rev & 0x0f) == 0x0a) { snd_ad1848_out(chip, AD1848_MISC_INFO, 0x40); for (i = 0; i < 16; ++i) { if (snd_ad1848_in(chip, i) != snd_ad1848_in(chip, i + 16)) { - chip->hardware = AD1848_HW_CMI8330; + chip->hardware = WSS_HW_CMI8330; break; } } @@ -782,7 +778,8 @@ static int snd_ad1848_playback_open(stru struct snd_pcm_runtime *runtime = substream->runtime; int err; - if ((err = snd_ad1848_open(chip, AD1848_MODE_PLAY)) < 0) + err = snd_ad1848_open(chip, WSS_MODE_PLAY); + if (err < 0) return err; chip->playback_substream = substream; runtime->hw = snd_ad1848_playback; @@ -798,7 +795,8 @@ static int snd_ad1848_capture_open(struc struct snd_pcm_runtime *runtime = substream->runtime; int err; - if ((err = snd_ad1848_open(chip, AD1848_MODE_CAPTURE)) < 0) + err = snd_ad1848_open(chip, WSS_MODE_RECORD); + if (err < 0) return err; chip->capture_substream = substream; runtime->hw = snd_ad1848_capture; @@ -812,7 +810,7 @@ static int snd_ad1848_playback_close(str { struct snd_wss *chip = snd_pcm_substream_chip(substream); - chip->mode &= ~AD1848_MODE_PLAY; + chip->mode &= ~WSS_MODE_PLAY; chip->playback_substream = NULL; snd_ad1848_close(chip); return 0; @@ -822,7 +820,7 @@ static int snd_ad1848_capture_close(stru { struct snd_wss *chip = snd_pcm_substream_chip(substream); - chip->mode &= ~AD1848_MODE_CAPTURE; + chip->mode &= ~WSS_MODE_RECORD; chip->capture_substream = NULL; snd_ad1848_close(chip); return 0; @@ -850,11 +848,16 @@ static int snd_ad1848_dev_free(struct sn static const char *snd_ad1848_chip_id(struct snd_wss *chip) { switch (chip->hardware) { - case AD1848_HW_AD1847: return "AD1847"; - case AD1848_HW_AD1848: return "AD1848"; - case AD1848_HW_CS4248: return "CS4248"; - case AD1848_HW_CMI8330: return "CMI8330/C3D"; - default: return "???"; + case WSS_HW_AD1847: + return "AD1847"; + case WSS_HW_AD1848: + return "AD1848"; + case WSS_HW_CS4248: + return "CS4248"; + case WSS_HW_CMI8330: + return "CMI8330/C3D"; + default: + return "???"; } } @@ -903,9 +906,9 @@ int snd_ad1848_create(struct snd_card *c chip->dma1 = dma; chip->dma2 = dma; - if (hardware == AD1848_HW_THINKPAD) { + if (hardware == WSS_HW_THINKPAD) { chip->thinkpad_flag = 1; - chip->hardware = AD1848_HW_DETECT; /* reset */ + chip->hardware = WSS_HW_DETECT; /* reset */ snd_ad1848_thinkpad_twiddle(chip, 1); } @@ -1214,6 +1217,20 @@ static const DECLARE_TLV_DB_SCALE(db_sca static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); +#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .name = xname, \ + .index = xindex, \ + .type = AD1848_MIX_SINGLE, \ + .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ + .tlv = xtlv } + +#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ +{ .name = xname, \ + .index = xindex, \ + .type = AD1848_MIX_DOUBLE, \ + .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ + .tlv = xtlv } + static struct ad1848_mix_elem snd_ad1848_controls[] = { AD1848_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), AD1848_DOUBLE_TLV("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, diff -urp linux-alsa/sound/isa/cmi8330.c linux-2.6.25/sound/isa/cmi8330.c --- linux-alsa/sound/isa/cmi8330.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/cmi8330.c 2008-07-08 21:43:51.000000000 +0200 @@ -465,12 +465,12 @@ static int __devinit snd_cmi8330_probe(s wssport[dev] + 4, wssirq[dev], wssdma[dev], - AD1848_HW_DETECT, + WSS_HW_DETECT, &acard->wss)) < 0) { snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } - if (acard->wss->hardware != AD1848_HW_CMI8330) { + if (acard->wss->hardware != WSS_HW_CMI8330) { snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); return -ENODEV; } diff -urp linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.25/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-08 18:06:22.000000000 +0200 @@ -771,7 +771,7 @@ static int __devinit snd_opti9xx_probe(s #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, - AD1848_HW_DETECT, &codec)) < 0) + WSS_HW_DETECT, &codec)) < 0) return error; if ((error = snd_ad1848_pcm(codec, 0, &pcm)) < 0) return error; diff -urp linux-alsa/sound/isa/sc6000.c linux-2.6.25/sound/isa/sc6000.c --- linux-alsa/sound/isa/sc6000.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/sc6000.c 2008-07-08 18:06:22.000000000 +0200 @@ -549,7 +549,7 @@ static int __devinit snd_sc6000_probe(st goto err_unmap2; err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, - AD1848_HW_DETECT, &chip); + WSS_HW_DETECT, &chip); if (err < 0) goto err_unmap2; card->private_data = chip; diff -urp linux-alsa/sound/isa/sgalaxy.c linux-2.6.25/sound/isa/sgalaxy.c --- linux-alsa/sound/isa/sgalaxy.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/sgalaxy.c 2008-07-08 21:43:51.000000000 +0200 @@ -265,7 +265,7 @@ static int __devinit snd_sgalaxy_probe(s if ((err = snd_ad1848_create(card, wssport[dev] + 4, xirq, xdma1, - AD1848_HW_DETECT, &chip)) < 0) + WSS_HW_DETECT, &chip)) < 0) goto _err; card->private_data = chip; ---------------------------------------------------------------------- Partyjka w Chinczyka? Graj >>> http://link.interia.pl/f1e67

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[alsa-devel] [PATCH] ALSA: ens1370: SRC stands for Sample Rate Converter
by Rene Herman 17 Jul '08

17 Jul '08

Trivial...

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