From 8e8b2d676f3f7c1246b108793fb5690e6c6fcd26 Mon Sep 17 00:00:00 2001 From: Eero Nurkkala Date: Mon, 12 Oct 2009 08:41:59 +0300 Subject: [PATCH 01/17] 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(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8de6f9dec4a2..d89f6dc00908 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; } From 4b7348a15972274eb16182d10987f69da3e95719 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Oct 2009 18:25:23 +0200 Subject: [PATCH 02/17] 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(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c08ca660daba..9b1cff83497f 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, 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: [PATCH 03/17] 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(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad2a97e..0a1b2f64bbee 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 * From b214f11fb92713fbb07d8c1f62dd1aa8077b56c9 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sat, 24 Oct 2009 00:06:48 +0200 Subject: [PATCH 04/17] 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(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 2dee9839be86..653a362425df 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" From 97609458ce972180172ae2cec0483451820e6a41 Mon Sep 17 00:00:00 2001 From: Wu Zhangjin Date: Thu, 15 Oct 2009 10:22:54 +0800 Subject: [PATCH 05/17] ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency SND_CS5535AUDIO is available on Loongson(MIPS compatible) family machines, and checked it with ARCH=x86_64, no relative compiling warnings & errors, so, remove the platform dependency directly. Reported-by: rixed@happyleptic.org Acked-by: Andres Salomon Signed-off-by: Wu Zhangjin Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index fb5ee3cc3968..75c602b5b132 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -259,7 +259,6 @@ config SND_CS5530 config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" - depends on X86 && !X86_64 select SND_PCM select SND_AC97_CODEC help From b71207e9dc044b30d8b5d7f1c2290ba14563f05c Mon Sep 17 00:00:00 2001 From: Stas Sergeev Date: Fri, 30 Oct 2009 11:51:24 +0100 Subject: [PATCH 06/17] ALSA: pcsp - Fix nforce workaround The attached patch fixes the problems introduced in this commit: http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=eea0579fc85e64e9f05361d5aacf496fe7a151aa - Fix nForce workaround by honouring the pointer_update var - Revert "ns" to u64, as per the hrtimer API - Revert to the zero-delay timer startup, since I can't reproduce any problem with it (please, give me the hint!) Signed-off-by: Stas Sergeev Signed-off-by: Takashi Iwai --- sound/drivers/pcsp/pcsp_lib.c | 67 +++++++++++++++++---------------- sound/drivers/pcsp/pcsp_mixer.c | 2 +- 2 files changed, 36 insertions(+), 33 deletions(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 84cc2658c05b..e1145ac6e908 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -39,25 +39,20 @@ static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0); /* write the port and returns the next expire time in ns; * called at the trigger-start and in hrtimer callback */ -static unsigned long pcsp_timer_update(struct hrtimer *handle) +static u64 pcsp_timer_update(struct snd_pcsp *chip) { unsigned char timer_cnt, val; u64 ns; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; - struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); unsigned long flags; if (chip->thalf) { outb(chip->val61, 0x61); chip->thalf = 0; - if (!atomic_read(&chip->timer_active)) - return 0; return chip->ns_rem; } - if (!atomic_read(&chip->timer_active)) - return 0; substream = chip->playback_substream; if (!substream) return 0; @@ -88,24 +83,17 @@ static unsigned long pcsp_timer_update(struct hrtimer *handle) return ns; } -enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +static void pcsp_pointer_update(struct snd_pcsp *chip) { - struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); struct snd_pcm_substream *substream; - int periods_elapsed, pointer_update; size_t period_bytes, buffer_bytes; - unsigned long ns; + int periods_elapsed; unsigned long flags; - pointer_update = !chip->thalf; - ns = pcsp_timer_update(handle); - if (!ns) - return HRTIMER_NORESTART; - /* update the playback position */ substream = chip->playback_substream; if (!substream) - return HRTIMER_NORESTART; + return; period_bytes = snd_pcm_lib_period_bytes(substream); buffer_bytes = snd_pcm_lib_buffer_bytes(substream); @@ -134,6 +122,26 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (periods_elapsed) tasklet_schedule(&pcsp_pcm_tasklet); +} + +enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +{ + struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + int pointer_update; + u64 ns; + + if (!atomic_read(&chip->timer_active) || !chip->playback_substream) + return HRTIMER_NORESTART; + + pointer_update = !chip->thalf; + ns = pcsp_timer_update(chip); + if (!ns) { + printk(KERN_WARNING "PCSP: unexpected stop\n"); + return HRTIMER_NORESTART; + } + + if (pointer_update) + pcsp_pointer_update(chip); hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); @@ -142,8 +150,6 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) static int pcsp_start_playing(struct snd_pcsp *chip) { - unsigned long ns; - #if PCSP_DEBUG printk(KERN_INFO "PCSP: start_playing called\n"); #endif @@ -159,11 +165,7 @@ static int pcsp_start_playing(struct snd_pcsp *chip) atomic_set(&chip->timer_active, 1); chip->thalf = 0; - ns = pcsp_timer_update(&pcsp_chip.timer); - if (!ns) - return -EIO; - - hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL); + hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); return 0; } @@ -232,21 +234,22 @@ static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); -#if PCSP_DEBUG - printk(KERN_INFO "PCSP: prepare called, " - "size=%zi psize=%zi f=%zi f1=%i\n", - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - snd_pcm_lib_buffer_bytes(substream) / - snd_pcm_lib_period_bytes(substream), - substream->runtime->periods); -#endif pcsp_sync_stop(chip); chip->playback_ptr = 0; chip->period_ptr = 0; chip->fmt_size = snd_pcm_format_physical_width(substream->runtime->format) >> 3; chip->is_signed = snd_pcm_format_signed(substream->runtime->format); +#if PCSP_DEBUG + printk(KERN_INFO "PCSP: prepare called, " + "size=%zi psize=%zi f=%zi f1=%i fsize=%i\n", + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + snd_pcm_lib_buffer_bytes(substream) / + snd_pcm_lib_period_bytes(substream), + substream->runtime->periods, + chip->fmt_size); +#endif return 0; } diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 199b03377142..903bc846763f 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -72,7 +72,7 @@ static int pcsp_treble_put(struct snd_kcontrol *kcontrol, if (treble != chip->treble) { chip->treble = treble; #if PCSP_DEBUG - printk(KERN_INFO "PCSP: rate set to %i\n", PCSP_RATE()); + printk(KERN_INFO "PCSP: rate set to %li\n", PCSP_RATE()); #endif changed = 1; } From db32f99816f7cbe61c1f75c1560655a3bf52488a Mon Sep 17 00:00:00 2001 From: peer chen Date: Thu, 15 Oct 2009 16:37:47 +0800 Subject: [PATCH 07/17] 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(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c9ad182e1b4b..e340792f6cb3 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 }, From 4b3be6afa4ab8b3fdce39df68bad71f8b85164de Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 17 Oct 2009 08:33:22 +0200 Subject: [PATCH 08/17] ALSA: sound: Move dereference after NULL test and drop unnecessary NULL tests In pcm.c, if the NULL test on pcm is needed, then the dereference should be after the NULL test. In dummy.c and ali5451.c, the context of the calls to snd_card_dummy_new_mixer and snd_ali_free_voice show that dummy and pvoice, respectively cannot be NULL. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 5 +++-- sound/drivers/dummy.c | 2 -- sound/pci/ali5451/ali5451.c | 2 +- 3 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 0c1440121c22..c69c60b2a48a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -953,11 +953,12 @@ static int snd_pcm_dev_register(struct snd_device *device) struct snd_pcm_substream *substream; struct snd_pcm_notify *notify; char str[16]; - struct snd_pcm *pcm = device->device_data; + struct snd_pcm *pcm; struct device *dev; - if (snd_BUG_ON(!pcm || !device)) + if (snd_BUG_ON(!device || !device->device_data)) return -ENXIO; + pcm = device->device_data; mutex_lock(®ister_mutex); err = snd_pcm_add(pcm); if (err) { diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 6ba066c41d2e..146ef00f94a3 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -808,8 +808,6 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) unsigned int idx; int err; - if (snd_BUG_ON(!dummy)) - return -EINVAL; spin_lock_init(&dummy->mixer_lock); strcpy(card->mixername, "Dummy Mixer"); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index b458d208720b..aaf4da68969c 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -973,7 +973,7 @@ static void snd_ali_free_voice(struct snd_ali * codec, void *private_data; snd_ali_printk("free_voice: channel=%d\n",pvoice->number); - if (pvoice == NULL || !pvoice->use) + if (!pvoice->use) return; snd_ali_clear_voices(codec, pvoice->number, pvoice->number); spin_lock_irq(&codec->voice_alloc); From e8e0929d7290cab7c5b1a3e5f5f54f73daf38038 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sat, 17 Oct 2009 08:33:47 +0200 Subject: [PATCH 09/17] ALSA: sound/parisc: Move dereference after NULL test If the NULL test on h is needed in snd_harmony_mixer_init, then the dereference should be after the NULL test. Actually, there is a sequence of calls: snd_harmony_create, then snd_harmony_pcm_init, and then snd_harmony_mixer_init. snd_harmony_create initializes h, but may indeed leave it as NULL. There was no NULL test at the beginning of snd_harmony_pcm_init, so I have added one. The NULL test in snd_harmony_mixer_init is then not necessary, but in case the ordering of the calls changes, I have left it, and moved the dereference after it. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/parisc/harmony.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index e924492df21d..f47f9e226b08 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -624,6 +624,9 @@ snd_harmony_pcm_init(struct snd_harmony *h) struct snd_pcm *pcm; int err; + if (snd_BUG_ON(!h)) + return -EINVAL; + harmony_disable_interrupts(h); err = snd_pcm_new(h->card, "harmony", 0, 1, 1, &pcm); @@ -865,11 +868,12 @@ snd_harmony_mixer_reset(struct snd_harmony *h) static int __devinit snd_harmony_mixer_init(struct snd_harmony *h) { - struct snd_card *card = h->card; + struct snd_card *card; int idx, err; if (snd_BUG_ON(!h)) return -EINVAL; + card = h->card; strcpy(card->mixername, "Harmony Gain control interface"); for (idx = 0; idx < HARMONY_CONTROLS; idx++) { From 3702b082281929cf1bdf14f67eb0619aab58b496 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Sat, 24 Oct 2009 12:59:35 +0100 Subject: [PATCH 10/17] ALSA: snd-usb-caiaq: Missing lock around use of buffer positions Fix a race which causes snd_pcm_update_hw_ptr_pos() to report a bug. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 121af0644fd9..e76017cd5acf 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -269,16 +269,22 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) { int index = sub->number; struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + snd_pcm_uframes_t ptr; + + spin_lock(&dev->spinlock); if (dev->input_panic || dev->output_panic) - return SNDRV_PCM_POS_XRUN; + ptr = SNDRV_PCM_POS_XRUN; if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_out_buf_pos[index]); else - return bytes_to_frames(sub->runtime, + ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); + + spin_unlock(&dev->spinlock); + return ptr; } /* operators for both playback and capture */ From ac9dd9d384b018f1e1c5a9a2686ab5605ce55818 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Sat, 24 Oct 2009 12:59:36 +0100 Subject: [PATCH 11/17] ALSA: snd-usb-caiaq: Lock on stream start/unpause Fix a bug which can result in white noise from the driver after stream start or unpause. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index e76017cd5acf..86b2c3b92df5 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -62,10 +62,14 @@ static void activate_substream(struct snd_usb_caiaqdev *dev, struct snd_pcm_substream *sub) { + spin_lock(&dev->spinlock); + if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) dev->sub_playback[sub->number] = sub; else dev->sub_capture[sub->number] = sub; + + spin_unlock(&dev->spinlock); } static void From 467cc1692036909ee0a723ce633fc4a53d72fd9a Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Sat, 24 Oct 2009 12:59:37 +0100 Subject: [PATCH 12/17] ALSA: snd-usb-caiaq: Bump version number to 1.3.20 Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 83e6c1312d47..a3f02dd97440 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 1.3.19"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," From 3d00941371a765779c4e3509214c7e5793cce1fe Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 22 Oct 2009 09:04:09 +0200 Subject: [PATCH 13/17] sound: via82xx: deactivate DXS controls of inactive streams Activate the DXS volume controls only when the corresponding stream is being used. This makes the behaviour consistent with the other drivers that have per-stream volume controls. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 59 +++++++++++++++++++++++++++++++++++++++------ 1 file changed, 52 insertions(+), 7 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 91683a349035..8a332d2f615c 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -386,6 +386,7 @@ struct via82xx { struct snd_pcm *pcms[2]; struct snd_rawmidi *rmidi; + struct snd_kcontrol *dxs_controls[4]; struct snd_ac97_bus *ac97_bus; struct snd_ac97 *ac97; @@ -1216,9 +1217,9 @@ static int snd_via82xx_pcm_open(struct via82xx *chip, struct viadev *viadev, /* - * open callback for playback on via686 and via823x DSX + * open callback for playback on via686 */ -static int snd_via82xx_playback_open(struct snd_pcm_substream *substream) +static int snd_via686_playback_open(struct snd_pcm_substream *substream) { struct via82xx *chip = snd_pcm_substream_chip(substream); struct viadev *viadev = &chip->devs[chip->playback_devno + substream->number]; @@ -1229,6 +1230,32 @@ static int snd_via82xx_playback_open(struct snd_pcm_substream *substream) return 0; } +/* + * open callback for playback on via823x DXS + */ +static int snd_via8233_playback_open(struct snd_pcm_substream *substream) +{ + struct via82xx *chip = snd_pcm_substream_chip(substream); + struct viadev *viadev; + unsigned int stream; + int err; + + viadev = &chip->devs[chip->playback_devno + substream->number]; + if ((err = snd_via82xx_pcm_open(chip, viadev, substream)) < 0) + return err; + stream = viadev->reg_offset / 0x10; + if (chip->dxs_controls[stream]) { + chip->playback_volume[stream][0] = 0; + chip->playback_volume[stream][1] = 0; + chip->dxs_controls[stream]->vd[0].access &= + ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, + &chip->dxs_controls[stream]->id); + } + return 0; +} + /* * open callback for playback on via823x multi-channel */ @@ -1302,10 +1329,26 @@ static int snd_via82xx_pcm_close(struct snd_pcm_substream *substream) return 0; } +static int snd_via8233_playback_close(struct snd_pcm_substream *substream) +{ + struct via82xx *chip = snd_pcm_substream_chip(substream); + struct viadev *viadev = substream->runtime->private_data; + unsigned int stream; + + stream = viadev->reg_offset / 0x10; + if (chip->dxs_controls[stream]) { + chip->dxs_controls[stream]->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, + &chip->dxs_controls[stream]->id); + } + return snd_via82xx_pcm_close(substream); +} + /* via686 playback callbacks */ static struct snd_pcm_ops snd_via686_playback_ops = { - .open = snd_via82xx_playback_open, + .open = snd_via686_playback_open, .close = snd_via82xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_via82xx_hw_params, @@ -1331,8 +1374,8 @@ static struct snd_pcm_ops snd_via686_capture_ops = { /* via823x DSX playback callbacks */ static struct snd_pcm_ops snd_via8233_playback_ops = { - .open = snd_via82xx_playback_open, - .close = snd_via82xx_pcm_close, + .open = snd_via8233_playback_open, + .close = snd_via8233_playback_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_via82xx_hw_params, .hw_free = snd_via82xx_hw_free, @@ -1709,8 +1752,9 @@ static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { .device = 0, /* .subdevice set later */ .name = "PCM Playback Volume", - .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_INACTIVE, .info = snd_via8233_dxs_volume_info, .get = snd_via8233_dxs_volume_get, .put = snd_via8233_dxs_volume_put, @@ -1948,6 +1992,7 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) err = snd_ctl_add(chip->card, kctl); if (err < 0) return err; + chip->dxs_controls[i] = kctl; } } } From a1bf808849f25a4d668f81415ecebb2da9fecf8e Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 1 Nov 2009 18:32:29 -0500 Subject: [PATCH 14/17] 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(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b1cff83497f..148734d16132 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 */ From 0d488234fd857aae07f1c56467bbf58f1a859753 Mon Sep 17 00:00:00 2001 From: Dominik Brodowski Date: Sat, 24 Oct 2009 21:43:03 +0200 Subject: [PATCH 15/17] 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(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 7dea74b71cf1..64b859925c0b 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 7445cc8a47d3..1492744ad67f 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; From 23aebca486429b74c35b41ac5cac7ce97609fd6a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Nov 2009 14:10:59 +0100 Subject: [PATCH 16/17] 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(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1c8eb4518ce0..fd9a2f67edf2 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 146ef00f94a3..252e04ce602f 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); From ad87c64f00e01a694bf90bddc2b4a6c90796d13c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Nov 2009 14:23:15 +0100 Subject: [PATCH 17/17] 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(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 148734d16132..ff20048504b6 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);