From 8e8b2d676f3f7c1246b108793fb5690e6c6fcd26 Mon Sep 17 00:00:00 2001 From: Eero Nurkkala Date: Mon, 12 Oct 2009 08:41:59 +0300 Subject: ASoC: Serialize access to dapm_power_widgets() Access to damp_power_widgets() is assumed to be single-threaded. Concurrent accesses to dapm_power_widgets() may result in unpredictable behavior. Calls from: close_delayed_work() soc_codec_close() soc_pcm_prepare() soc_suspend() soc_resume_deferred() to snd_soc_dapm_stream_event() do not have the codec->mutex taken to cover the call to dapm_power_widgets(). Thus, take the mutex in these paths also to assure single-threaded use of dapm_power_widgets(). Signed-off-by: Eero Nurkkala Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8de6f9dec4a..d89f6dc0090 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2072,9 +2072,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, } } } - mutex_unlock(&codec->mutex); dapm_power_widgets(codec, event); + mutex_unlock(&codec->mutex); dump_dapm(codec, __func__); return 0; } -- cgit v1.2.3-70-g09d2 From 4b7348a15972274eb16182d10987f69da3e95719 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Oct 2009 18:25:23 +0200 Subject: ALSA: hda - Fix capture source checks for ALC662/663 codecs The ALC662/663 parser calls wrongly alc880_auto_create_input_ctls() to check the capture source selections. This should be alc882, instead. Reference: Novell bnc#546918 http://bugzilla.novell.com/show_bug.cgi?id=546918 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c08ca660dab..9b1cff83497 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17374,7 +17374,7 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, /* create playback/capture controls for input pins */ #define alc662_auto_create_input_ctls \ - alc880_auto_create_input_ctls + alc882_auto_create_input_ctls static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, -- cgit v1.2.3-70-g09d2 From 02a06d3042e208cb74369838b178ca9512192be4 Mon Sep 17 00:00:00 2001 From: Barry Song <21cnbao@gmail.com> Date: Fri, 16 Oct 2009 18:13:38 +0800 Subject: ASoC: Fix possible codec_dai->ops NULL pointer problems Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc. access the ops field in these DAIs, panic will happen. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad2a97..0a1b2f64bbe 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -834,6 +834,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume_device); #define soc_resume NULL #endif +static struct snd_soc_dai_ops null_dai_ops = { +}; + static void snd_soc_instantiate_card(struct snd_soc_card *card) { struct platform_device *pdev = container_of(card->dev, @@ -877,6 +880,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ac97 = 1; } + for (i = 0; i < card->num_links; i++) { + if (!card->dai_link[i].codec_dai->ops) + card->dai_link[i].codec_dai->ops = &null_dai_ops; + } + /* If we have AC97 in the system then don't wait for the * codec. This will need revisiting if we have to handle * systems with mixed AC97 and non-AC97 parts. Only check for @@ -2329,9 +2337,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } -static struct snd_soc_dai_ops null_dai_ops = { -}; - /** * snd_soc_register_dai - Register a DAI with the ASoC core * -- cgit v1.2.3-70-g09d2 From b214f11fb92713fbb07d8c1f62dd1aa8077b56c9 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sat, 24 Oct 2009 00:06:48 +0200 Subject: ASoC: Amstrad Delta: add info about the line discipline requirement to Kconfig help text I thought it could be usefull to add some information on how to get the device fully supported by loading a line discipline on the modem line. Signed-off-by: Janusz Krzysztofik Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2dee9839be8..653a362425d 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA select SND_OMAP_SOC_MCBSP select SND_SOC_CX20442 help - Say Y if you want to add support for SoC audio on Amstrad Delta. + Say Y if you want to add support for SoC audio device connected to + a handset and a speakerphone found on Amstrad E3 (Delta) videophone. + + Note that in order to get those devices fully supported, you have to + build the kernel with standard serial port driver included and + configured for at least 4 ports. Then, from userspace, you must load + a line discipline #19 on the modem (ttyS3) serial line. The simplest + way to achieve this is to install util-linux-ng and use the included + ldattach utility. This can be started automatically from udev, + a simple rule like this one should do the trick (it does for me): + ACTION=="add", KERNEL=="controlC0", \ + RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" -- cgit v1.2.3-70-g09d2 From f8a3ae6c84e60a3f35f573ea592b8fe00dd367ab Mon Sep 17 00:00:00 2001 From: Kumar Gala Date: Fri, 16 Oct 2009 07:21:38 +0000 Subject: powerpc: Minor cleanup to sound/ppc/Kconfig We can replace PPC32 || PPC64 as a dependancy with just PPC as all powerpc platforms (32-bit and 64-bit) define PPC now. Signed-off-by: Kumar Gala Signed-off-by: Benjamin Herrenschmidt --- sound/ppc/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index bd2338ab2ce..0519c60f5be 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -2,7 +2,7 @@ menuconfig SND_PPC bool "PowerPC sound devices" - depends on PPC64 || PPC32 + depends on PPC default y help Support for sound devices specific to PowerPC architectures. -- cgit v1.2.3-70-g09d2 From e3d8024891dbfec6cf36c9b76177650f48118462 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Tue, 22 Sep 2009 16:48:56 +0100 Subject: ARM: S3C: Add info for supporting circular DMA buffers The S3C64XX DMA implementation will work a lot better with the ability to enqueue circular buffers as the hardware can do it's own linked-list management. Add a function s3c_dma_has_circular() to show that the system can do this and a flag for the channel. Update the s3c24xx/s3c64xx I2S DMA code to deal with this. Signed-off-by: Ben Dooks Signed-off-by: Ben Dooks Acked-by: Mark Brown --- arch/arm/mach-s3c2410/include/mach/dma.h | 7 +++++++ arch/arm/mach-s3c6400/include/mach/dma.h | 5 +++++ sound/soc/s3c24xx/s3c24xx-pcm.c | 17 +++++++++++++++-- 3 files changed, 27 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/arch/arm/mach-s3c2410/include/mach/dma.h b/arch/arm/mach-s3c2410/include/mach/dma.h index c3a2629e0de..92e2687009e 100644 --- a/arch/arm/mach-s3c2410/include/mach/dma.h +++ b/arch/arm/mach-s3c2410/include/mach/dma.h @@ -110,6 +110,8 @@ enum s3c2410_dma_loadst { * waiting for reloads */ #define S3C2410_DMAF_AUTOSTART (1<<1) /* auto-start if buffer queued */ +#define S3C2410_DMAF_CIRCULAR (1 << 2) /* no circular dma support */ + /* dma buffer */ struct s3c2410_dma_buf; @@ -194,4 +196,9 @@ struct s3c2410_dma_chan { typedef unsigned long dma_device_t; +static inline bool s3c_dma_has_circular(void) +{ + return false; +} + #endif /* __ASM_ARCH_DMA_H */ diff --git a/arch/arm/mach-s3c6400/include/mach/dma.h b/arch/arm/mach-s3c6400/include/mach/dma.h index 1067619f0ba..004edab2395 100644 --- a/arch/arm/mach-s3c6400/include/mach/dma.h +++ b/arch/arm/mach-s3c6400/include/mach/dma.h @@ -68,6 +68,11 @@ static __inline__ int s3c_dma_has_circular(void) #define S3C2410_DMAF_CIRCULAR (1 << 0) +static inline bool s3c_dma_has_circular(void) +{ + return false; +} + #include #endif /* __ASM_ARCH_IRQ_H */ diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 5cbbdc80fde..1f35c6fcf5f 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -75,11 +75,19 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; + unsigned int limit; int ret; pr_debug("Entered %s\n", __func__); - while (prtd->dma_loaded < prtd->dma_limit) { + if (s3c_dma_has_circular()) { + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + } else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { unsigned long len = prtd->dma_period; pr_debug("dma_loaded: %d\n", prtd->dma_loaded); @@ -123,7 +131,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING) { + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { prtd->dma_loaded--; s3c24xx_pcm_enqueue(substream); } @@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, printk(KERN_ERR "failed to get dma channel\n"); return ret; } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); } s3c2410_dma_set_buffdone_fn(prtd->params->channel, -- cgit v1.2.3-70-g09d2 From db32f99816f7cbe61c1f75c1560655a3bf52488a Mon Sep 17 00:00:00 2001 From: peer chen Date: Thu, 15 Oct 2009 16:37:47 +0800 Subject: ALSA: hda_intel: Add the Linux device ID for NVIDIA HDA controller Add the generic device ID for NVIDIA HDA controller. Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c9ad182e1b4..e340792f6cb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2674,6 +2674,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, -- cgit v1.2.3-70-g09d2 From a1bf808849f25a4d668f81415ecebb2da9fecf8e Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 1 Nov 2009 18:32:29 -0500 Subject: ALSA: hda: Use quirk mask for Dell Inspiron Mini9/Vostro A90 using ALC268 BugLink: https://bugs.launchpad.net/bugs/368629 We should use a quirk mask for these Dell Inspiron Mini9s and Vostro A90s, as the model=dell quirk appears to enable audio on them. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b1cff83497..148734d1613 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12602,7 +12602,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), - SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0, + "Dell Inspiron Mini9/Vostro A90", ALC268_DELL), /* almost compatible with toshiba but with optional digital outs; * auto-probing seems working fine */ -- cgit v1.2.3-70-g09d2 From 0d488234fd857aae07f1c56467bbf58f1a859753 Mon Sep 17 00:00:00 2001 From: Dominik Brodowski Date: Sat, 24 Oct 2009 21:43:03 +0200 Subject: ALSA: pcmcia: use dynamic debug infrastructure, deprecate CS_CHECK (sound) Convert PCMCIA drivers to use the dynamic debug infrastructure, instead of requiring manual settings of PCMCIA_DEBUG. Also, remove all usages of the CS_CHECK macro and replace them with proper Linux style calling and return value checking. The extra error reporting may be dropped, as the PCMCIA core already complains about any (non-driver-author) errors. Signed-off-by: Dominik Brodowski Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 21 ++++++++++++--------- sound/pcmcia/vx/vxpocket.c | 21 ++++++++++++--------- 2 files changed, 24 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 7dea74b71cf..64b859925c0 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -217,20 +217,25 @@ static void snd_pdacf_detach(struct pcmcia_device *link) * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int pdacf_config(struct pcmcia_device *link) { struct snd_pdacf *pdacf = link->priv; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "pdacf_config called\n"); link->conf.ConfigIndex = 0x5; - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0) goto failed; @@ -238,8 +243,6 @@ static int pdacf_config(struct pcmcia_device *link) link->dev_node = &pdacf->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7445cc8a47d..1492744ad67 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -213,14 +213,11 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq * configuration callback */ -#define CS_CHECK(fn, ret) \ -do { last_fn = (fn); if ((last_ret = (ret)) != 0) goto cs_failed; } while (0) - static int vxpocket_config(struct pcmcia_device *link) { struct vx_core *chip = link->priv; struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - int last_fn, last_ret; + int ret; snd_printdd(KERN_DEBUG "vxpocket_config called\n"); @@ -235,9 +232,17 @@ static int vxpocket_config(struct pcmcia_device *link) strcpy(chip->card->driver, vxp440_hw.name); } - CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); - CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); + ret = pcmcia_request_io(link, &link->io); + if (ret) + goto failed; + + ret = pcmcia_request_irq(link, &link->irq); + if (ret) + goto failed; + + ret = pcmcia_request_configuration(link, &link->conf); + if (ret) + goto failed; chip->dev = &handle_to_dev(link); snd_card_set_dev(chip->card, chip->dev); @@ -248,8 +253,6 @@ static int vxpocket_config(struct pcmcia_device *link) link->dev_node = &vxp->node; return 0; -cs_failed: - cs_error(link, last_fn, last_ret); failed: pcmcia_disable_device(link); return -ENODEV; -- cgit v1.2.3-70-g09d2 From 23aebca486429b74c35b41ac5cac7ce97609fd6a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Nov 2009 14:10:59 +0100 Subject: ALSA: dummy - Fix descriptions of pcm_substreams parameter Now up to 128 substreams are supported. Reported-by: Adrian Bridgett Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- sound/drivers/dummy.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb4518ce..fd9a2f67edf 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -522,7 +522,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. pcm_devs - Number of PCM devices assigned to each card (default = 1, up to 4) pcm_substreams - Number of PCM substreams assigned to each PCM - (default = 8, up to 16) + (default = 8, up to 128) hrtimer - Use hrtimer (=1, default) or system timer (=0) fake_buffer - Fake buffer allocations (default = 1) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 146ef00f94a..252e04ce602 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -165,7 +165,7 @@ MODULE_PARM_DESC(enable, "Enable this dummy soundcard."); module_param_array(pcm_devs, int, NULL, 0444); MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver."); module_param_array(pcm_substreams, int, NULL, 0444); -MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver."); +MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); module_param(fake_buffer, bool, 0444); -- cgit v1.2.3-70-g09d2 From ad87c64f00e01a694bf90bddc2b4a6c90796d13c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Nov 2009 14:23:15 +0100 Subject: ALSA: hda - Don't check invalid HP pin alc_automute_pin() might be called even if any HP pin is defined, and it will result in verbs with NID=0. This patch adds a check for the validity of HP widget before issuing any verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 148734d1613..ff20048504b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -965,6 +965,8 @@ static void alc_automute_pin(struct hda_codec *codec) unsigned int nid = spec->autocfg.hp_pins[0]; int i; + if (!nid) + return; pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); -- cgit v1.2.3-70-g09d2 From 5bdaaada16363d64e10ae081755d1a8d392429f2 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Wed, 4 Nov 2009 07:57:45 +0100 Subject: ALSA: hda - Enable GPIO control for mute LED on HP systems This patch enables GPIO to control mute LED indicator on the HP systems with the special string in BIOS and applies it with the correct polarity on HP B-series systems. It also restores configuration of the pin intended as the second Headphone on HP B-series systems but configured as something else in the BIOS to pass MS DTM. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 68 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 68 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 66c0876bf73..b513eba2d2f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -1693,6 +1694,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, "HP dv4-1222nr", STAC_HP_DV4_1222NR), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x1720, + "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, @@ -4730,6 +4733,26 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } +static int hp_bseries_system(u32 subsystem_id) +{ + switch (subsystem_id) { + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c1722: + case 0x103c1723: + case 0x103c1724: + case 0x103c1725: + case 0x103c1726: + case 0x103c1727: + case 0x103c1728: + case 0x103c1729: + return 1; + } + return 0; +} + /* * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) @@ -4754,6 +4777,11 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, else spec->gpio_data |= spec->gpio_led; /* white */ + if (hp_bseries_system(codec->subsystem_id)) { + /* LED state is inverted on these systems */ + spec->gpio_data ^= spec->gpio_led; + } + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); @@ -5243,6 +5271,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; + unsigned int pin_cfg; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5426,6 +5455,45 @@ again: break; } + if (hp_bseries_system(codec->subsystem_id)) { + pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); + if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || + get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || + get_defcfg_device(pin_cfg) == AC_JACK_HP_OUT) { + /* It was changed in the BIOS to just satisfy MS DTM. + * Lets turn it back into slaved HP + */ + pin_cfg = (pin_cfg & (~AC_DEFCFG_DEVICE)) + | (AC_JACK_HP_OUT << + AC_DEFCFG_DEVICE_SHIFT); + pin_cfg = (pin_cfg & (~(AC_DEFCFG_DEF_ASSOC + | AC_DEFCFG_SEQUENCE))) + | 0x1f; + snd_hda_codec_set_pincfg(codec, 0x0f, pin_cfg); + } + } + + if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { + const struct dmi_device *dev = NULL; + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, + NULL, dev))) { + if (strcmp(dev->name, "HP_Mute_LED_1")) { + switch (codec->vendor_id) { + case 0x111d7608: + spec->gpio_led = 0x01; + break; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + spec->gpio_led = 0x08; + break; + } + break; + } + } + } + #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { spec->gpio_mask |= spec->gpio_led; -- cgit v1.2.3-70-g09d2 From 798a8a15011e88cc63dbbb15728b42572c152093 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Wed, 4 Nov 2009 10:11:07 +0000 Subject: ALSA: hda - Add OLPC XO-1.5 PCI ID The XO-1.5 laptop now has a unique subvendor/subproduct ID, which can be used to automatically select the correct CXT5066 configuration. Signed-off-by: Daniel Drake Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3fbbc8c01e7..6479e65858d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2325,6 +2325,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_LAPTOP), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), + SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), {} }; -- cgit v1.2.3-70-g09d2 From 7e6c3989af5baee999ef9a4424e85938cba8d34a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 4 Nov 2009 21:03:46 -0500 Subject: ALSA: intel8x0: Mute External Amplifier by default for another Sony model BugLink: https://bugs.launchpad.net/bugs/474972 This Sony model needs External Amplifier muted for audible playback, so make sure we set the inv_eapd quirk. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 754867ed478..aac20fb4aad 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1948,6 +1948,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP xw4200", /* AD1981B*/ .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x104d, + .subdevice = 0x8144, + .name = "Sony", + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x104d, .subdevice = 0x8197, -- cgit v1.2.3-70-g09d2 From f702cf463e1308fbb0c1faa9f3d8e3fa9cb5630f Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Wed, 4 Nov 2009 16:04:52 -0800 Subject: sound: Use KERN_WARNING instead of KERN_WARN, which does not exist Reported-by: Andrew Lyon Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/oss/sb_common.c | 4 ++-- sound/oss/sb_ess.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 77d0e5efda7..ce4db49291f 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -157,7 +157,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } @@ -177,7 +177,7 @@ static void sb_intr (sb_devc *devc) break; default: - /* printk(KERN_WARN "Sound Blaster: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "Sound Blaster: Unexpected interrupt\n"); */ ; } } diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 180e95c87e3..51a3d381a59 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -782,7 +782,7 @@ printk(KERN_INFO "FKS: ess_handle_channel %s irq_mode=%d\n", channel, irq_mode); break; default:; - /* printk(KERN_WARN "ESS: Unexpected interrupt\n"); */ + /* printk(KERN_WARNING "ESS: Unexpected interrupt\n"); */ } } -- cgit v1.2.3-70-g09d2 From 78987bdc4e41a425ac113c2c51474f0368fe653a Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Thu, 5 Nov 2009 09:22:30 -0800 Subject: ALSA: hda, move hp_bseries_system Function hp_bseries_system() is always used, outside of CONFIG_ boundaries/controls, so move it. sound/pci/hda/patch_sigmatel.c:5458: error: implicit declaration of function 'hp_bseries_system' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b513eba2d2f..8eb6508cd99 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4668,6 +4668,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_bseries_system(u32 subsystem_id) +{ + switch (subsystem_id) { + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c1722: + case 0x103c1723: + case 0x103c1724: + case 0x103c1725: + case 0x103c1726: + case 0x103c1727: + case 0x103c1728: + case 0x103c1729: + return 1; + } + return 0; +} + #ifdef CONFIG_PROC_FS static void stac92hd_proc_hook(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) @@ -4733,26 +4753,6 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } -static int hp_bseries_system(u32 subsystem_id) -{ - switch (subsystem_id) { - case 0x103c307e: - case 0x103c307f: - case 0x103c3080: - case 0x103c3081: - case 0x103c1722: - case 0x103c1723: - case 0x103c1724: - case 0x103c1725: - case 0x103c1726: - case 0x103c1727: - case 0x103c1728: - case 0x103c1729: - return 1; - } - return 0; -} - /* * using power check for controlling mute led of HP notebooks * check for mute state only on Speakers (nid = 0x10) -- cgit v1.2.3-70-g09d2 From 4d187fb830a7aa8afb471124abe41b04caf49401 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Wed, 21 Oct 2009 23:10:03 +0200 Subject: ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1 After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c, omap_pcm_prepare() unconditionally calls: omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); Current implementation of those two functions found in arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at all, so they both end with BUG() on that machine. That results in ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta. The patch corrects the problem by not calling those two functions when run on OMAP1 class based machines. Created against linux-2.6.32-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 5735945788b..6a829eef2a4 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) else omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); - omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); - omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); + if (!(cpu_class_is_omap1())) { + omap_set_dma_src_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + omap_set_dma_dest_burst_mode(prtd->dma_ch, + OMAP_DMA_DATA_BURST_16); + } return 0; } -- cgit v1.2.3-70-g09d2 From 6fc786d5034ed7ce2d43c459211137de6d99dd28 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Fri, 6 Nov 2009 18:00:24 +0900 Subject: ASoC: S3C64XX I2S: Enable audio-bus clock Added the missing clk_enable after acquiring the 'audio-bus' clock. Signed-off-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 3c06c401d0f..105a77eeded 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -220,6 +220,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) goto err; } + clk_enable(i2s->iis_cclk); + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); if (ret) goto err_clk; -- cgit v1.2.3-70-g09d2 From 70edc800a39327174d3244f9226ce8cacd01dc91 Mon Sep 17 00:00:00 2001 From: Thomas Gleixner Date: Fri, 6 Nov 2009 22:41:29 +0000 Subject: sound: Replace old style lock initializer SPIN_LOCK_UNLOCKED is deprecated. Use __SPIN_LOCK_UNLOCKED instead. Signed-off-by: Thomas Gleixner Signed-off-by: Takashi Iwai --- sound/oss/dmasound/dmasound_core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 793b7f47843..3f3c3f71db4 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -219,7 +219,9 @@ static int shared_resources_initialised; * Mid level stuff */ -struct sound_settings dmasound = { .lock = SPIN_LOCK_UNLOCKED }; +struct sound_settings dmasound = { + .lock = __SPIN_LOCK_UNLOCKED(dmasound.lock) +}; static inline void sound_silence(void) { -- cgit v1.2.3-70-g09d2 From f495088210c8b9e20791d995a8210170c68d2deb Mon Sep 17 00:00:00 2001 From: Julian Anastasov Date: Fri, 6 Nov 2009 23:44:53 +0200 Subject: ALSA: usb-audio: fix combine_word problem Fix combine_word problem where first octet is not read properly. The only affected place seems to be the INPUT_TERMINAL type. Before now, sound controls can be created with the output terminal's name which is a fallback mechanism used only for unknown input terminal types. For example, Line can wrongly appear as Speaker. After the change it should appear as Line. The side effect of this change can be that users can expect the wrong control name in their scripts or programs while now we return the correct one. Probably, these defines should use get_unaligned_le16 and friends. Signed-off-by: Julian Anastasov Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 8e7f78941ba..e9a3a9dca15 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -210,7 +210,7 @@ struct snd_usb_midi_endpoint_info { /* */ -#define combine_word(s) ((*s) | ((unsigned int)(s)[1] << 8)) +#define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) -- cgit v1.2.3-70-g09d2 From 1a6969788ef2d5bc3169eee59def6b267182f136 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 7 Nov 2009 09:49:04 +0100 Subject: ALSA: hda - Don't initialize CORB/RIRB for single_cmd mode So far, CORB/RIRB still remains even if the driver is switched to the single_cmd mode. The specification says that this should be disabled, but I hoped this isn't the case; indeed most devices worked together with CORB/RIRB. However, Poulsbo (US15W) seems problematic with this setup, and it requires to disable CORB/RIRB when single_cmd is used. Now this patch disables CORB/RIRB initialization when the single_cmd mode is used. Also the unsolicited event is disabled because it can't work without RIRB. Reported-and-tested-by: Troy Kisky Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e340792f6cb..6517f589d01 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -722,9 +722,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, chip->last_cmd[addr]); chip->single_cmd = 1; bus->response_reset = 0; - /* re-initialize CORB/RIRB */ + /* release CORB/RIRB */ azx_free_cmd_io(chip); - azx_init_cmd_io(chip); + /* disable unsolicited responses */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_UNSOL); return -1; } @@ -865,7 +866,9 @@ static int azx_reset(struct azx *chip) } /* Accept unsolicited responses */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) | ICH6_GCTL_UNSOL); + if (!chip->single_cmd) + azx_writel(chip, GCTL, azx_readl(chip, GCTL) | + ICH6_GCTL_UNSOL); /* detect codecs */ if (!chip->codec_mask) { @@ -980,7 +983,8 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - azx_init_cmd_io(chip); + if (!chip->single_cmd) + azx_init_cmd_io(chip); /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); -- cgit v1.2.3-70-g09d2 From f37325a956f0ee4356793da7d93c699a25b21d92 Mon Sep 17 00:00:00 2001 From: Ben Hutchings Date: Sat, 7 Nov 2009 22:13:39 +0000 Subject: ALSA: snd-aica: declare MODULE_FIRMWARE Signed-off-by: Ben Hutchings Signed-off-by: Takashi Iwai --- sound/sh/aica.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 583a3693df7..a0df401ebb9 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -49,6 +49,7 @@ MODULE_AUTHOR("Adrian McMenamin "); MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); +MODULE_FIRMWARE("aica_firmware.bin"); /* module parameters */ #define CARD_NAME "AICA" -- cgit v1.2.3-70-g09d2 From 8579d2d7779d7ff41ea2a0183015e0e5038f1043 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:09:38 +0200 Subject: sound: rawmidi: fix double init when opening MIDI device with O_APPEND Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 moved the substream initialization code to where it would be executed every time the substream is opened. This had the consequence that any further opening would drop and leak the data in the existing buffer, and that the device driver's open callback would be called multiple times, unexpectedly. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c0adc14c91f..3071e6f5801 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -266,17 +266,19 @@ static int open_substream(struct snd_rawmidi *rmidi, { int err; - err = snd_rawmidi_runtime_create(substream); - if (err < 0) - return err; - err = substream->ops->open(substream); - if (err < 0) - return err; - substream->opened = 1; - if (substream->use_count++ == 0) + if (substream->use_count == 0) { + err = snd_rawmidi_runtime_create(substream); + if (err < 0) + return err; + err = substream->ops->open(substream); + if (err < 0) + return err; + substream->opened = 1; substream->active_sensing = 0; - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) - substream->append = 1; + if (mode & SNDRV_RAWMIDI_LFLG_APPEND) + substream->append = 1; + } + substream->use_count++; rmidi->streams[substream->stream].substream_opened++; return 0; } -- cgit v1.2.3-70-g09d2 From 16fb109644b5644e42ececeff644514de6f4bd03 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:10:16 +0200 Subject: sound: rawmidi: fix checking of O_APPEND when opening MIDI device Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 dropped the check that a substream must already have been opened with O_APPEND to be able to open it a second time. This would make it possible for a substream to be switched to append mode, which would mean that non-atomic writes would fail unexpectedly. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 3071e6f5801..091405385e1 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -248,7 +248,8 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, list_for_each_entry(substream, &s->substreams, list) { if (substream->opened) { if (stream == SNDRV_RAWMIDI_STREAM_INPUT || - !(mode & SNDRV_RAWMIDI_LFLG_APPEND)) + !(mode & SNDRV_RAWMIDI_LFLG_APPEND) || + !substream->append) continue; } if (subdevice < 0 || subdevice == substream->number) { -- cgit v1.2.3-70-g09d2 From b7fe750fcceda4fa6bef399b0e2812562728ea82 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 21 Oct 2009 09:11:43 +0200 Subject: sound: rawmidi: fix MIDI device O_APPEND error handling Commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a in 2.6.30 broke the error handling code in rawmidi_open_priv(). If only the output substream of a RawMIDI device has been opened and if this device is then opened with O_RDWR | O_APPEND and if the initialization of the input substream fails (either because of low memory or because the device driver's open callback fails), then the runtime structure of the already open output substream will be freed and all following writes through the first handle will cause snd_rawmidi_write() to use the NULL runtime pointer. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 091405385e1..70d6f25ba52 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -272,8 +272,10 @@ static int open_substream(struct snd_rawmidi *rmidi, if (err < 0) return err; err = substream->ops->open(substream); - if (err < 0) + if (err < 0) { + snd_rawmidi_runtime_free(substream); return err; + } substream->opened = 1; substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) @@ -300,27 +302,27 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, SNDRV_RAWMIDI_STREAM_INPUT, mode, &sinput); if (err < 0) - goto __error; + return err; } if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { err = assign_substream(rmidi, subdevice, SNDRV_RAWMIDI_STREAM_OUTPUT, mode, &soutput); if (err < 0) - goto __error; + return err; } if (sinput) { err = open_substream(rmidi, sinput, mode); if (err < 0) - goto __error; + return err; } if (soutput) { err = open_substream(rmidi, soutput, mode); if (err < 0) { if (sinput) close_substream(rmidi, sinput, 0); - goto __error; + return err; } } @@ -328,13 +330,6 @@ static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, rfile->input = sinput; rfile->output = soutput; return 0; - - __error: - if (sinput && sinput->runtime) - snd_rawmidi_runtime_free(sinput); - if (soutput && soutput->runtime) - snd_rawmidi_runtime_free(soutput); - return err; } /* called from sound/core/seq/seq_midi.c */ -- cgit v1.2.3-70-g09d2 From 5e08fe570c2dbabb5015c37049eb9a451e55c890 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 14 Nov 2009 14:37:19 +0100 Subject: ALSA: ice1724 - Fix section mismatch in prodigy_hd2_resume() Remove invlid __devinit prefix from the suspend callback. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/prodigy_hifi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index c75515f5be6..6a9fee3ee78 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice) } #ifdef CONFIG_PM -static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice) +static int prodigy_hd2_resume(struct snd_ice1712 *ice) { /* initialize ak4396 codec and restore previous mixer volumes */ struct prodigy_hifi_spec *spec = ice->spec; -- cgit v1.2.3-70-g09d2