May 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Donation of ARM CubieTruck to anyone interested in developing
by Alexey Eromenko 30 May '14

30 May '14

Hello, My name is Alexey, and I'm interested in having ARM CubieTruck hardware well supported by desktop (KDE) use. CubieTruck is an ARM mini-PC aka "Liliputer", based on AllWinner A20 SoC + 2 GB RAM + 8 GB flash + VGA (!) port + SATA + Ethernet + WiFi, making it good enough for a real Debian desktop. CubieTruck http://cubietruck.com/collections/frontpage/products/cubietruck-cubieboard3… My goal: It should have a working 2D, HD Video and 3D graphics as well as audio and networking. Some open-source drivers exist: *2D - xf86-video-fbturbo - link: https://github.com/ssvb/xf86-video-fbturbo *Hardware Video Decoder - "CedarX" link: http://linux-sunxi.org/Reverse_Engineering/Cedar_Status (should play H.264 and WebM/VP8 Full HD 1080p video) *3D - open-source Lima or closed-source Mali drivers. WiFi + Bluetooth - status unknown. (WiFi firmware is probably closed source) If you can - please help stabilize those experimental drivers, especially sound driver is semi-broken. I can donate one ARM CubieTruck sample to interested people. You'll need a proof of past patches to any open-source project or community involvement. -- -Alexey Eromenko "Technologov". Open-source community member. Israel.

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Re: [alsa-devel] [PATCH] ALSA: firewire: Fix dependency on PCM and rawmidi
by Takashi Iwai 30 May '14

30 May '14

At Fri, 30 May 2014 21:15:58 +0900, Takashi Sakamoto wrote: > > (May 30 2014 17:22), Takashi Iwai wrote: > > Now snd-firewire-lib supports rawmidi in addition to PCM, thus we need > > to give a proper dependency. For fixing and simplification, move the > > selections of SND_PCM and SND_RAWMIDI into SND_FIREWIRE_LIB section. > > Then each driver doesn't have to select them but only > > SND_FIREWIRE_LIB. > > > > Reported-by: Jim Davis <jim.epost(a)gmail.com> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > Tested-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> > > Thanks for this patch. I've received this bug from kbuild test backend > but I'm not so good at Kbuild system itself and have been investigating > how the Kconfig should be. > > If possible, the dependency of firewire-lib should be switchable because > some of firewire devices don't support MIDI interfaces. For example, > iSight doesn't. So there may be needs not to build MIDI modules. > > But for this idea, MIDI functionalities of each drivers also need to be > disable. As a result, this idea makes the codes complicated a bit. So I > agree with this patch. In that case, we'd need either splitting to yet another helper module only for rawmidi (e.g. snd-firewire-rawmidi), or introduce a Kconfig and select it from the card driver, then build conditionally into snd-firewire-libs. OTOH, the current form has a clear merit of simplicity, indeed. So, I applied the patch now. thanks, Takashi > > > Thanks > > Takashi Sakamoto > o-takashi(a)sakamocchi.jp > > > --- > > sound/firewire/Kconfig | 12 ++---------- > > 1 file changed, 2 insertions(+), 10 deletions(-) > > > > diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig > > index 9f363fac7688..775ef2efc296 100644 > > --- a/sound/firewire/Kconfig > > +++ b/sound/firewire/Kconfig > > @@ -9,12 +9,12 @@ if SND_FIREWIRE && FIREWIRE > > > > config SND_FIREWIRE_LIB > > tristate > > - depends on SND_PCM > > + select SND_PCM > > + select SND_RAWMIDI > > > > config SND_DICE > > tristate "DICE-based DACs (EXPERIMENTAL)" > > select SND_HWDEP > > - select SND_PCM > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for many DACs based on the DICE > > @@ -28,7 +28,6 @@ config SND_DICE > > > > config SND_FIREWIRE_SPEAKERS > > tristate "FireWire speakers" > > - select SND_PCM > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for the Griffin FireWave Surround > > @@ -39,7 +38,6 @@ config SND_FIREWIRE_SPEAKERS > > > > config SND_ISIGHT > > tristate "Apple iSight microphone" > > - select SND_PCM > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for the front and rear microphones > > @@ -50,8 +48,6 @@ config SND_ISIGHT > > > > config SND_SCS1X > > tristate "Stanton Control System 1 MIDI" > > - select SND_PCM > > - select SND_RAWMIDI > > select SND_FIREWIRE_LIB > > help > > Say Y here to include support for the MIDI ports of the Stanton > > @@ -64,8 +60,6 @@ config SND_SCS1X > > config SND_FIREWORKS > > tristate "Echo Fireworks board module support" > > select SND_FIREWIRE_LIB > > - select SND_RAWMIDI > > - select SND_PCM > > select SND_HWDEP > > help > > Say Y here to include support for FireWire devices based > > @@ -82,8 +76,6 @@ config SND_FIREWORKS > > config SND_BEBOB > > tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" > > select SND_FIREWIRE_LIB > > - select SND_RAWMIDI > > - select SND_PCM > > select SND_HWDEP > > help > > Say Y here to include support for FireWire devices based >

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[alsa-devel] [PATCH] ALSA: hda/analog - Fix silent output on ASUS A8JN
by Takashi Iwai 30 May '14

30 May '14

ASUS A8JN with AD1986A codec seems following the normal EAPD in the normal order (0 = off, 1 = on) unlike other machines with AD1986A. Apply the workaround used for Toshiba laptop that showed the same problem. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=75041 Cc: <stable(a)vger.kernel.org> [3.11+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 40ba06eb44af..06275f8807a8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -332,6 +332,7 @@ static const struct hda_fixup ad1986a_fixups[] = { static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK), -- 1.9.3

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[alsa-devel] [PATCH] sound: remove checks for CONFIG_SND_DEBUG_ROM
by Paul Bolle 30 May '14

30 May '14

Checks for CONFIG_SND_DEBUG_ROM were added in v2.5.5 but a Kconfig symbol SND_DEBUG_ROM was never added. These checks have always evaluated to false. Remove them and the printk()s they hide. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Untested. Perhaps one is expected to define this debugging macro by hand, somehow. In that case it would be nice to drop the CONFIG_ prefix. sound/isa/gus/interwave.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 5abbbe477d16..ad55e5cb8e94 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -442,17 +442,11 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) for (bank_pos = 0; bank_pos < 16L * 1024L * 1024L; bank_pos += 4L * 1024L * 1024L) { for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); -#ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave); -#endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ csum = 0; for (i = 0; i < sizeof(struct rom_hdr); i++) csum += snd_gf1_peek(gus, bank_pos + i); -#ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM checksum = 0x%x (computed)\n", csum); -#endif if (csum != 0) continue; /* not valid rom */ gus->gf1.rom_banks++; -- 1.9.3

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[alsa-devel] [PATCH] sound: remove checks for CONFIG_BCM_CS4297A_CSWARM
by Paul Bolle 30 May '14

30 May '14

Checks for CONFIG_BCM_CS4297A_CSWARM were added in v2.6.11. The related Kconfig symbol was never added so these checks always evaluated to true. Remove them. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Untested. sound/oss/swarm_cs4297a.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index f851fd0e199c..a33e8ce8085b 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -2625,15 +2625,12 @@ static int __init cs4297a_init(void) u32 pwr, id; mm_segment_t fs; int rval; -#ifndef CONFIG_BCM_CS4297A_CSWARM u64 cfg; int mdio_val; -#endif CS_DBGOUT(CS_INIT | CS_FUNCTION, 2, printk(KERN_INFO "cs4297a: cs4297a_init_module()+ \n")); -#ifndef CONFIG_BCM_CS4297A_CSWARM mdio_val = __raw_readq(KSEG1 + A_MAC_REGISTER(2, R_MAC_MDIO)) & (M_MAC_MDIO_DIR|M_MAC_MDIO_OUT); @@ -2659,7 +2656,6 @@ static int __init cs4297a_init(void) __raw_writeq(mdio_val | M_MAC_GENC, KSEG1+A_MAC_REGISTER(2, R_MAC_MDIO)); /* Give the codec some time to finish resetting (start the bit clock) */ udelay(100); -#endif if (!(s = kzalloc(sizeof(struct cs4297a_state), GFP_KERNEL))) { CS_DBGOUT(CS_ERROR, 1, printk(KERN_ERR -- 1.9.3

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[alsa-devel] [PATCH] ALSA: firewire: Fix dependency on PCM and rawmidi
by Takashi Iwai 30 May '14

30 May '14

Now snd-firewire-lib supports rawmidi in addition to PCM, thus we need to give a proper dependency. For fixing and simplification, move the selections of SND_PCM and SND_RAWMIDI into SND_FIREWIRE_LIB section. Then each driver doesn't have to select them but only SND_FIREWIRE_LIB. Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/firewire/Kconfig | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 9f363fac7688..775ef2efc296 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -9,12 +9,12 @@ if SND_FIREWIRE && FIREWIRE config SND_FIREWIRE_LIB tristate - depends on SND_PCM + select SND_PCM + select SND_RAWMIDI config SND_DICE tristate "DICE-based DACs (EXPERIMENTAL)" select SND_HWDEP - select SND_PCM select SND_FIREWIRE_LIB help Say Y here to include support for many DACs based on the DICE @@ -28,7 +28,6 @@ config SND_DICE config SND_FIREWIRE_SPEAKERS tristate "FireWire speakers" - select SND_PCM select SND_FIREWIRE_LIB help Say Y here to include support for the Griffin FireWave Surround @@ -39,7 +38,6 @@ config SND_FIREWIRE_SPEAKERS config SND_ISIGHT tristate "Apple iSight microphone" - select SND_PCM select SND_FIREWIRE_LIB help Say Y here to include support for the front and rear microphones @@ -50,8 +48,6 @@ config SND_ISIGHT config SND_SCS1X tristate "Stanton Control System 1 MIDI" - select SND_PCM - select SND_RAWMIDI select SND_FIREWIRE_LIB help Say Y here to include support for the MIDI ports of the Stanton @@ -64,8 +60,6 @@ config SND_SCS1X config SND_FIREWORKS tristate "Echo Fireworks board module support" select SND_FIREWIRE_LIB - select SND_RAWMIDI - select SND_PCM select SND_HWDEP help Say Y here to include support for FireWire devices based @@ -82,8 +76,6 @@ config SND_FIREWORKS config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB - select SND_RAWMIDI - select SND_PCM select SND_HWDEP help Say Y here to include support for FireWire devices based -- 1.9.3

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[alsa-devel] Testing rewindability of a pcm
by Alexander E. Patrakov 30 May '14

30 May '14

Hello. [David: if I misrepresent the result of our discussion, please correct me.] On the pulseaudio-discuss list, there was a thread started by me: http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-April/020449.… The initial patch attempted to work around known brokenness of rewind handling in certain ALSA plugins by keeping a whitelist of ALSA plugin types known to implement full or almost-full rewinds absolutely correctly, and falling back to the low-latency no-rewinds mode on others (such as extplug, which only pretends to support rewinds). While David accepted that it would be a good idea to use a not-so-high latency (he proposed ~150-200 ms) if rewinds are not available, keeping a plugin whitelist inside PulseAudio code was shot down as a workaround for an ALSA bug. The alternative proposal is to fix ALSA so that it correctly reports rewindability, and use the reported data, which looks like a good idea. However, there remain some disagreements on the details how ALSA needs to be fixed, that's why this e-mail. David suggests that PulseAudio should use the snd_pcm_rewindable() function when all plugins are fixed to either implement rewinds correctly or to return 0 if they don't actually support rewinds. And for ioplug and extplug, supporting rewinds is not really an option, because the backend library (which can be an AC3 or DTS encoder) does not necessarily have a rewindable API. <offtopic><troll>And we don't even have a rewindable resampler implementation!</troll></offtopic> While fixing snd_pcm_rewindable() is indeed a valid task that I take for some plugins, I have a question. """ Should PulseAudio indeed use a complex dance with snd_pcm_rewindable() to get the essentially-static bit of information whether a given PCM supports full rewinds? Isn't there a better solution? """ The problem is that the information should be used to set the maximum allowed latency of the sink: a PCM that supports full rewinds can be set to 2 seconds of latency, while for a PCM that doesn't support rewinds, 150-200 ms is more appropriate. In other words, ideally, one bit of information about the future full-rewind support should be available at the server startup without any complex dance with test audio streams. Ideally, even before setting hw_params, but "just after" also seems to be acceptable. And at this stage (before or just after setting hw_params), snd_pcm_rewindable() is useless, because it returns 0 even for the hw plugin which does support full rewinds (rightfully, because the card initially is just about to underrun). Therefore, if the solution is indeed to use snd_pcm_rewindable(), then it is needed to write some test data to the pcm, and only then test for rewindability. Alternatives are below, please help choosing: 1) Use a test stream and call snd_pcm_rewindable() when there is a non-zero amount of data in the buffer: I don't like it. Too complex dance, with side effects, to get one bit of essentially-static information that the plugin knows for sure even before setting hw_params. 2) Use snd_pcm_forwardable() with an empty buffer just after setting hw_params: works (once the implementation in ioplug/extplug is fixed to return 0), doesn't require API additions, but David doesn't like it. He thinks that the sets of PCMs that support forwarding and rewinding may be different (here I disagree, but can't provide arguments). Additionally, forwarding the application pointer without writing anything and rewinding over that beforehand looks like undefined behaviour (which looks like a valid argument, but I am not 100% sure, and I am not actually forwarding, I am just asking how much I can forward). 3) Add a new function to ALSA that gets this bit of static information for a given pcm handle, use it. Drawbacks: a new function is needed, and it's a bad idea to send a patch that adds a new public ALSA API function without discussing here first (that's why the e-mail). 4) Write your own alternative here if you have one. -- Alexander E. Patrakov

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[alsa-devel] [PATCH] ASoC: wm8804: Allow control of master clock divider in PLL generation
by Charles Keepax 29 May '14

29 May '14

From: Daniel Matuschek <info(a)crazy-audio.com> WM8804 can run with PLL frequencies of 256xfs and 128xfs for most sample rates. At 192kHz only 128xfs is supported. The existing driver selects 128xfs automatically for some lower samples rates. By using an additional mclk_div divider, it is now possible to control the behaviour. This allows using 256xfs PLL frequency on all sample rates up to 96kHz. It should allow lower jitter and better signal quality. The behavior has to be controlled by the sound card driver, because some sample frequency share the same setting. e.g. 192kHz and 96kHz use 24.576MHz master clock. The only difference is the MCLK divider. Signed-off-by: Daniel Matuschek <daniel(a)matuschek.net> Tested-by: Florian Meier <florian.meier(a)koalo.de> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Hi, Looks like the version of this patch that was sent in didn't apply on the ASoC tree which is probably why it seems to have been missed. Not sure whether this was an email client fail or the patch was generated against the wrong tree but as it is a useful patch I have taken the liberty of fixing it up (code is exactly the same) and resending. Thanks, Charles sound/soc/codecs/wm8804.c | 17 ++++++++++++++--- sound/soc/codecs/wm8804.h | 4 ++++ 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index bbcad9f..d96e596 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -63,6 +63,7 @@ struct wm8804_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; + int mclk_div; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -318,7 +319,7 @@ static struct { #define FIXED_PLL_SIZE ((1ULL << 22) * 10) static int pll_factors(struct pll_div *pll_div, unsigned int target, - unsigned int source) + unsigned int source, unsigned int mclk_div) { u64 Kpart; unsigned long int K, Ndiv, Nmod, tmp; @@ -330,7 +331,8 @@ static int pll_factors(struct pll_div *pll_div, unsigned int target, */ for (i = 0; i < ARRAY_SIZE(post_table); i++) { tmp = target * post_table[i].div; - if (tmp >= 90000000 && tmp <= 100000000) { + if ((tmp >= 90000000 && tmp <= 100000000) && + (mclk_div == post_table[i].mclkdiv)) { pll_div->freqmode = post_table[i].freqmode; pll_div->mclkdiv = post_table[i].mclkdiv; target *= post_table[i].div; @@ -387,8 +389,12 @@ static int wm8804_set_pll(struct snd_soc_dai *dai, int pll_id, } else { int ret; struct pll_div pll_div; + struct wm8804_priv *wm8804; - ret = pll_factors(&pll_div, freq_out, freq_in); + wm8804 = snd_soc_codec_get_drvdata(codec); + + ret = pll_factors(&pll_div, freq_out, freq_in, + wm8804->mclk_div); if (ret) return ret; @@ -452,6 +458,7 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { struct snd_soc_codec *codec; + struct wm8804_priv *wm8804; codec = dai->codec; switch (div_id) { @@ -459,6 +466,10 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, snd_soc_update_bits(codec, WM8804_PLL5, 0x30, (div & 0x3) << 4); break; + case WM8804_MCLK_DIV: + wm8804 = snd_soc_codec_get_drvdata(codec); + wm8804->mclk_div = div; + break; default: dev_err(dai->dev, "Unknown clock divider: %d\n", div_id); return -EINVAL; diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index 8ec14f5..e72d4f4 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -57,5 +57,9 @@ #define WM8804_CLKOUT_SRC_OSCCLK 4 #define WM8804_CLKOUT_DIV 1 +#define WM8804_MCLK_DIV 2 + +#define WM8804_MCLKDIV_256FS 0 +#define WM8804_MCLKDIV_128FS 1 #endif /* _WM8804_H */ -- 1.7.2.5

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[alsa-devel] [PATCH 1/3] Revert "ALSA: hda - drop def association and sequence from pinconf comparing"
by Hui Wang 29 May '14

29 May '14

This reverts commit c687200b9d4ef60042a50f7d942cfef120cc7bf1. Dropping the def association and sequence from pinconf comparing is a bit risky, It will introduce a greater risk of catching unwanted machines. And in addition, so far no BIOS experts give us an explicit answer whether it makes senses to compare these two fields or not. For safety reason, we revert this commit. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- This is the first attempt to move existing machines from old quirk table to hda_pin_quirk table. During the working, I met serveral cases that the same codec has different default pin conf value just because of default association difference, Please see the following two patches (0x10ec0255 and 0x10ec0668). sound/pci/hda/hda_auto_parser.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 3cf9137..b684c6e 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -844,8 +844,7 @@ static bool pin_config_match(struct hda_codec *codec, { for (; pins->nid; pins++) { u32 def_conf = snd_hda_codec_get_pincfg(codec, pins->nid); - u32 mask = 0xffffff00; - if ((pins->val & mask) != (def_conf & mask)) + if (pins->val != def_conf) return false; } return true; -- 1.8.1.2

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[alsa-devel] [patch] ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
by Dan Carpenter 29 May '14

29 May '14

ARRAY_SIZE() was intended here instead of sizeof(). The "bridgeco_freq_table" array holds integers so the original condition is never true. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 514c7c9..bc4f827 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -749,7 +749,7 @@ parse_stream_formation(u8 *buf, unsigned int len, if (buf[2] == bridgeco_freq_table[i]) break; } - if (i == sizeof(bridgeco_freq_table)) + if (i == ARRAY_SIZE(bridgeco_freq_table)) return -ENOSYS; /* Avoid double count by different entries for the same rate. */

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