From 6a84c234da06a4ac0c1b4c819b83cf264674c2d8 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sun, 28 Jun 2009 01:41:52 -0600 Subject: ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS. Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctly registered on the AC97 bus, which prevents thinks like the WM9712 touchscreen driver from getting probed. Tested against 2.6.31-rc1. Signed-off-by: Grant Likely Acked-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 5dbebf82249..5661876ee83 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S config SND_SOC_MPC5200_AC97 tristate "Freescale MPC5200 PSC in AC97 mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM - select AC97_BUS + select SND_SOC_AC97_BUS select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help -- cgit v1.2.3 From 40d9ec14e7e1f62d2379ecc1b5ee00ddfc2a5d0c Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sun, 28 Jun 2009 01:42:06 -0600 Subject: ASoC: remove BROKEN from Efika and pcm030 fabric drivers The needed spin_event_timeout() macro is now merged in from the powerpc tree, so these drivers are no longer broken. This reverts commit 0c0e09e21a9e7bc6ca54e06ef3d497255ca26383 (ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved) Tested against 2.6.31-rc1. Signed-off-by: Grant Likely Acked-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 5661876ee83..8cb65ccad35 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -41,7 +41,7 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE && BROKEN + depends on PPC_MPC5200_SIMPLE select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help @@ -50,7 +50,7 @@ config SND_MPC52xx_SOC_PCM030 config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA && BROKEN + depends on PPC_EFIKA select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help -- cgit v1.2.3 From 1bdd7419910c1506151e7b9e2d60c6980e015f76 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sun, 28 Jun 2009 00:21:05 +0200 Subject: ASoC: OMAP: fix OMAP1510 broken PCM pointer callback This patch tries to work around the problem of broken OMAP1510 PCM playback pointer calculation by replacing DMA function call that incorrectly tries to read the value form DMA hardware with a value computed locally from an already maintained variable omap_runtime_data.period_index. Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASoC driver. Based on linux-2.6-asoc.git v2.6.31-rc1. Signed-off-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 6454e15f7d2..84a1950880e 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ptr = omap_get_dma_src_pos(prtd->dma_ch); - else + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else if (!(cpu_is_omap1510())) { + ptr = omap_get_dma_src_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else + offset = prtd->period_index * runtime->period_size; - offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); if (offset >= runtime->buffer_size) offset = 0; -- cgit v1.2.3 From 1e1689536f346a431b748dc8ad9ac0828d2c065d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Jul 2009 08:34:32 +0200 Subject: ALSA: hda - Add missing static to patch_ca0110() Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 392d108c355..019ca7cb56d 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec) } -int patch_ca0110(struct hda_codec *codec) +static int patch_ca0110(struct hda_codec *codec) { struct ca0110_spec *spec; int err; -- cgit v1.2.3 From ff84847171508a3c76eb7e483204d1be7738729b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Jul 2009 18:08:01 +0200 Subject: ALSA: hda - Add quirk for HP 6930p Added a quirk model=laptop for HP 6930p (103c:30dc) with AD1984A codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 84cc49ca914..85e8618e849 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3966,6 +3966,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), -- cgit v1.2.3 From da9ff1f796e81976935407251815838bef9868d4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Jul 2009 18:23:26 +0100 Subject: ASoC: Only disable pxa2xx-i2s clocks if we enabled them The clock API can't cope with unbalanced enables and disables and we only enable in hw_params() but try to disable in shutdown. Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 4743e262895..6b8f655d1ad 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -167,6 +167,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); + dai->private_data = dai; pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -255,7 +256,10 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); - clk_disable(clk_i2s); + if (dai->private_data != NULL) { + clk_disable(clk_i2s); + dai->private_data = NULL; + } } } @@ -336,6 +340,7 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) return PTR_ERR(clk_i2s); pxa_i2s_dai.dev = &dev->dev; + pxa_i2s_dai.private_data = NULL; ret = snd_soc_register_dai(&pxa_i2s_dai); if (ret != 0) clk_put(clk_i2s); -- cgit v1.2.3 From 826390796d09444b93e1f957582f8970ddfd9b3d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Jul 2009 08:31:30 +0200 Subject: sound: virtuoso: fix Xonar D1/DX silence after resume When resuming, we better take the DACs out of the reset state before trying to use them. Reference: kernel bug #13599 http://bugzilla.kernel.org/show_bug.cgi?id=13599 Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bf971f7cfdc..6ebcb6bdd71 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip) static void xonar_d1_resume(struct oxygen *chip) { + oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); + msleep(1); cs43xx_init(chip); xonar_enable_output(chip); } -- cgit v1.2.3 From 563c2bf59d392357bcc1d99642933cc88c687964 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 2 Jul 2009 10:46:35 +0200 Subject: ALSA: snd_usb_caiaq: reparent sound device The sound device instance needs to be a child of the USB interface, not the USB device. Newer udev versions pay attention to that. Signed-off-by: Daniel Mack Reported-by: Lennart Poettering Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 0e5db719de2..de38108f0b2 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -349,7 +349,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) log("Unable to set up control system (ret=%d)\n", ret); } -static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) +static int create_card(struct usb_device *usb_dev, + struct usb_interface *intf, + struct snd_card **cardp) { int devnum; int err; @@ -374,7 +376,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), le16_to_cpu(usb_dev->descriptor.idProduct)); spin_lock_init(&dev->spinlock); - snd_card_set_dev(card, &usb_dev->dev); + snd_card_set_dev(card, &intf->dev); *cardp = card; return 0; @@ -461,7 +463,7 @@ static int __devinit snd_probe(struct usb_interface *intf, struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - ret = create_card(device, &card); + ret = create_card(device, intf, &card); if (ret < 0) return ret; -- cgit v1.2.3 From 3f5d3465be8f6e04f43d9b6d543fe28d4be07d78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Jul 2009 11:51:44 +0200 Subject: ALSA: usx2y - reparent sound device Fix the parent device to be the USB interface, not the USB device. A similiar commit like 563c2bf59d392357bcc1d99642933cc88c687964. Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 2 +- sound/usb/usx2y/usbusx2y.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index a5aae9d67f3..fd44946ce4b 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -514,7 +514,6 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) US122L(card)->chip.dev->bus->busnum, US122L(card)->chip.dev->devnum ); - snd_card_set_dev(card, &device->dev); *cardp = card; return 0; } @@ -531,6 +530,7 @@ static int us122l_usb_probe(struct usb_interface *intf, if (err < 0) return err; + snd_card_set_dev(card, &intf->dev); if (!us122l_create_card(card)) { snd_card_free(card); return -EINVAL; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 5ce0da23ee9..cb4bb8373ca 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -364,7 +364,6 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) 0,//us428(card)->usbmidi.ifnum, usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum ); - snd_card_set_dev(card, &device->dev); *cardp = card; return 0; } @@ -388,6 +387,7 @@ static int usX2Y_usb_probe(struct usb_device *device, err = usX2Y_create_card(device, &card); if (err < 0) return err; + snd_card_set_dev(card, &intf->dev); if ((err = usX2Y_hwdep_new(card, device)) < 0 || (err = snd_card_register(card)) < 0) { snd_card_free(card); -- cgit v1.2.3 From 099db17e66294b02814dee01c81d9abbbeece93e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Jul 2009 16:10:23 +0200 Subject: ALSA: hda - Add GPIO1 control at muting with HP laptops HP laptops with AD1984A codecs (at least mobile models) need to set GPIO1 appropriately to indicate the mute state. The BIOS checks this bit to judge whether the mute on or off is sent via F8 key. Without changing this bit, the BIOS can be confused and may toggle the mute wrongly. Reference: Novell bnc#515266 https://bugzilla.novell.com/show_bug.cgi?id=515266 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 85e8618e849..f795ee588cc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3734,9 +3734,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; +static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + int mute = (!ucontrol->value.integer.value[0] && + !ucontrol->value.integer.value[1]); + /* toggle GPIO1 according to the mute state */ + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + mute ? 0x02 : 0x0); + return ret; +} + static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), @@ -3857,6 +3878,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = { /* unsolicited event for pin-sense */ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ { } /* end */ }; -- cgit v1.2.3 From aa202455eec51699e44f658530728162cefa1307 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 15:00:54 +0200 Subject: ALSA: hda - Improve ASUS eeePC 1000 mixer The mixer elements created for ASUS eeePC 1000 with ALC269 aren't standard but strange words like "LineOut". Rename the element names to follow the standard one like "Headphone" and "Speaker". Also, split the volumes to each so that the virtual master can control them. The alc269_fujitsu_mixer is removed because it's now identical with the new eeepc mixer. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++------------------- 1 file changed, 5 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a8e58c483d..e661b21354b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12876,20 +12876,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -/* bind volumes of both NID 0x0c and 0x0d */ -static struct hda_bind_ctls alc269_epc_bind_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), - 0 - }, -}; - static struct snd_kcontrol_new alc269_eeepc_mixer[] = { - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol), - HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -12902,12 +12893,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { }; /* FSC amilo */ -static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol), - { } /* end */ -}; +#define alc269_fujitsu_mixer alc269_eeepc_mixer static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, -- cgit v1.2.3 From 022b466fc353d3dc7a152451144be656248666ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:03:30 +0200 Subject: ALSA: hda - Avoid invalid formats and rates with shared SPDIF Check whether formats and rates don't result in zero due to the restriction of SPDIF sharing. If any of them can be zero, disable the SPDIF sharing mode instead. Otherwise it will lead to a PCM configuration error. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2cedaa6..26d255de6be 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3470,10 +3470,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, } mutex_lock(&codec->spdif_mutex); if (mout->share_spdif) { - runtime->hw.rates &= mout->spdif_rates; - runtime->hw.formats &= mout->spdif_formats; - if (mout->spdif_maxbps < hinfo->maxbps) - hinfo->maxbps = mout->spdif_maxbps; + if ((runtime->hw.rates & mout->spdif_rates) && + (runtime->hw.formats & mout->spdif_formats)) { + runtime->hw.rates &= mout->spdif_rates; + runtime->hw.formats &= mout->spdif_formats; + if (mout->spdif_maxbps < hinfo->maxbps) + hinfo->maxbps = mout->spdif_maxbps; + } else { + mout->share_spdif = 0; + /* FIXME: need notify? */ + } } mutex_unlock(&codec->spdif_mutex); } -- cgit v1.2.3 From 70d321e6380f128096429d6e5b678f94ab0cef5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:06:45 +0200 Subject: ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback The PCM rates bit field may have been changed by the codec open callback. In that case, we need to reset rate_min and rate_max. So, simply call snd_pcm_lib_hw_rates() again after the codec open callback. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4e9ea708027..b36dc46615a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1454,6 +1454,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&chip->open_mutex); return err; } + snd_pcm_limit_hw_rates(runtime); spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; azx_dev->running = 0; -- cgit v1.2.3 From c470331e69bd54d11a9ea3c27a0e4ad783d02d6b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:10:23 +0200 Subject: ALSA: hda - Add sanity check in PCM open callback Add some sanity checks of struct snd_pcm_hardware fields in the PCM open callback of hda driver. This makes a bit easier to debug any PCM setup errors in the codec side. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b36dc46615a..1877d95d4aa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1464,6 +1464,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); + if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max)) + return -EINVAL; + if (snd_BUG_ON(!runtime->hw.formats)) + return -EINVAL; + if (snd_BUG_ON(!runtime->hw.rates)) + return -EINVAL; return 0; } -- cgit v1.2.3 From 954a973cab37ad5df3f87f08964166abd956cc17 Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Fri, 3 Jul 2009 20:56:05 +0200 Subject: sound: do not set DEVNAME for OSS devices Signed-off-by: Kay Sievers Signed-off-by: Takashi Iwai --- sound/sound_core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 12522e6913d..a41f8b127f4 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -10,6 +10,8 @@ #include #include #include +#include +#include #include #ifdef CONFIG_SOUND_OSS_CORE @@ -29,6 +31,8 @@ MODULE_LICENSE("GPL"); static char *sound_nodename(struct device *dev) { + if (MAJOR(dev->devt) == SOUND_MAJOR) + return NULL; return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev)); } @@ -104,7 +108,6 @@ module_exit(cleanup_soundcore); #include #include #include -#include #include #define SOUND_STEP 16 -- cgit v1.2.3