June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Indirect Callback with ack is called with no data
by Singaravelan Nallasellan 23 Jun '09

23 Jun '09

I am using indirect callback mechanism inside ack. I see that ack does not call my indirect callback pointer. Going through the code, I suspect that the appl_ptr is not being incremented and there is no data for the indirect callback to be called. After few acks, I get the indirect callback called. With aplay, I am able to play the wav file. But with flash, there is not data being sent to the driver. Please help me with any clues. Thanks in advance. Sing _________________________________________________________________ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor…

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Re: [alsa-devel] alc889
by Takashi Iwai 23 Jun '09

23 Jun '09

At Tue, 23 Jun 2009 00:05:42 +0200, Stefan Krastanov wrote: > > Hello, > > Thank you for your fast response, I appreciate it, knowing you don't have all > that free time. > > As I have said my hardware configuration is: laptop with alc889 soundcard > - two front > speakers > - one > sub-woofer > - one > internal microphone > - jacks > with jacksensing (tested on windows) - mic, line-in, headphones > - digital > IO > - > potentiometer for volume control > > I installed the latest snapshot using ubuntu's automated script (I manually > verified that it's the latest) and used model=auto. The results were the same > when I tried it few months ago: > - "Front" controls the front speakers OK > - "Front" also controls headphones problem > - "Headphones" slider does not control anything problem > - No control slider for the build-in sub-woofer problem > - microphones working (external and internal) OK > - potentiometer for volume control not working problem > - Line-in not tested, digital IO not tested not tested > - jack sensing on the headphones is not working problem > - jack sensing on microphone and line in not tested not tested > > Attached to the message you can find the output from alsa-info. Hm, how about model=fujitsu-pi2515 or model=fujitsu-xa3530 option? Just a wild guess from the PCI SSID, though. Takashi

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[alsa-devel] Noisy USB audio stream using MPlayer
by Eran Ben-Avi 23 Jun '09

23 Jun '09

Hi, I encountered some issue with my USB audio device and couldn't find any similar issue in previous mailing lists, any help will be appreciated. When playing an audio stream using MPlayer Ver1.0rc2-4.1.2 (tried different sample rates 44K, 48K and different files WAV , MP3) using my Creative SB0271 USB device on ARM based system(Marvell Kirkwood SoC) and kernel 2.6.22.18, I hear an interfering constant noise additionally to the original audio stream. At the beginning I heard this noise also when I used aplay(for WAV) but after I reinstalled the ALSA-LIB and ALSA-UTILS(version 1.0.14rc4), the aplay suddenly started to work smooth. I tested different USB audio device(AKM) but got same noisy results. Another piece of information I can provide is that increasing the CPU utilization eliminates the noise completely - maybe a timing issue ??? Any ideas ? Thanks, Eran.

2 3

[alsa-devel] [PATCH] ASoC: TWL4030: Correct bypass event for voice sidetone
by Lopez Cruz, Misael 23 Jun '09

23 Jun '09

Event for voice sidetone was being interpreted as an analog HiFi bypass event because VSTPGA register offset is less than ARXR2_APGA_CTL offset. Reordering the register checks allows to handle voice digital bypass event properly. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 20 +++++++++++++------- 1 files changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f916a9a..df42fa2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -712,7 +712,19 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + /* + * bypass_state[0:3] - analog HiFi bypass + * bypass_state[4] - analog voice bypass + * bypass_state[5] - digital voice bypass + * bypass_state[6:7] - digital HiFi bypass + */ + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +738,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) -- 1.5.4.3

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[alsa-devel] [PATCH] ASoC: TWL4030: Add AVADC Clock Priority
by Lopez Cruz, Misael 23 Jun '09

23 Jun '09

AVDAC clk priority allows to determine the path ADC must be connected when the codec is in option2 and both HiFi and Voice paths are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 780217a..42cdcce 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1036,6 +1036,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "HiFi high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1137,6 +1147,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), -- 1.5.4.3

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[alsa-devel] Nmode issues
by Dmitri Seletski 23 Jun '09

23 Jun '09

Hello Guys. I got upgraded my kernel to uname -a Linux dimko 2.6.30 #3 SMP PREEMPT Mon Jun 22 23:58:16 IST 2009 x86_64 AMD Athlon(tm) 64 X2 Dual Core Processor 6000+ AuthenticAMD GNU/Linux i was able to make files now!(previously i couldnt) make make install modprobe ath9k [ 941.365887] f71882fg: Not a Fintek device [ 941.365932] f71882fg: Found f71882fg chip at 0xa00, revision 32 [ 941.366116] f71882fg f71882fg.2560: Fan: 1 is in duty-cycle mode [ 941.366118] f71882fg f71882fg.2560: Fan: 2 is in duty-cycle mode [ 941.366119] f71882fg f71882fg.2560: Fan: 3 is in duty-cycle mode [ 941.366121] f71882fg f71882fg.2560: Fan: 4 is in duty-cycle mode [ 941.377478] k8temp 0000:00:18.3: Temperature readouts might be wrong - check erratum #141 [ 956.642074] IRQ 19/nvidia: IRQF_DISABLED is not guaranteed on shared IRQs [ 957.433234] IRQ 19/nvidia: IRQF_DISABLED is not guaranteed on shared IRQs [ 1415.180230] AV200 0000:09:04.0: PCI INT A -> Link[LN4A] -> GSI 16 (level, low) -> IRQ 16 [ 1509.674361] cfg80211: Calling CRDA to update world regulatory domain [ 1509.674399] general protection fault: 0000 [#1] PREEMPT SMP [ 1509.674411] last sysfs file: /sys/devices/pci0000:00/0000:00:04.1/usb2/2-1/2-1:1.0/host8/target8:0:0/8:0:0:0/block/sdd/sdd1/stat [ 1509.674418] CPU 1 [ 1509.674422] Modules linked in: cfg80211 rfkill_backport snd_seq snd_virtuoso snd_oxygen_lib snd_pcm snd_timer snd_page_alloc snd_mpu401_uart snd_rawmidi snd_seq_device snd soundcore k8temp f71882fg hwmon i2c_dev nvidia(P) i2c_core [ 1509.674464] Pid: 29571, comm: modprobe Tainted: P 2.6.30 #3 MS-7375 [ 1509.674470] RIP: 0010:[<ffffffff810710f1>] [<ffffffff810710f1>] module_put+0x31/0x80 [ 1509.674488] RSP: 0018:ffff880119cebf38 EFLAGS: 00010287 [ 1509.674493] RAX: 0000000000000001 RBX: ffffffffa0886d00 RCX: 0000000000000000 [ 1509.674498] RDX: ffff10013a92b6c0 RSI: 0000000000000000 RDI: 0000000000000001 [ 1509.674503] RBP: ffff880119cebf48 R08: 0000000000000000 R09: 0000000000000000 [ 1509.674508] R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 [ 1509.674513] R13: 00007f4dae114010 R14: 00000000006da7f0 R15: 00000000006da808 [ 1509.674519] FS: 00007f4dae1766f0(0000) GS:ffff88002803b000(0000) knlGS:0000000000000000 [ 1509.674524] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 1509.674529] CR2: 00007fb3a126d098 CR3: 000000010f527000 CR4: 00000000000006e0 [ 1509.674534] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 1509.674539] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 1509.674545] Process modprobe (pid: 29571, threadinfo ffff880119cea000, task ffff880119d0c380) [ 1509.674549] Stack: [ 1509.674552] ffffffffa0886d00 ffffffffa0886d00 ffff880119cebf78 ffffffff810730ce [ 1509.674560] 0000000000000000 00000000006da680 0000000000000000 00000000006da6e0 [ 1509.674569] 00000000006da680 ffffffff8100beeb 0000000000000246 00000000006da6c0 [ 1509.674578] Call Trace: [ 1509.674583] [<ffffffff810730ce>] sys_init_module+0x10e/0x200 [ 1509.674592] [<ffffffff8100beeb>] system_call_fastpath+0x16/0x1b [ 1509.674604] Code: 89 fb 48 83 ec 08 48 85 ff 74 44 bf 01 00 00 00 e8 25 d5 fc ff e8 d0 48 15 00 48 98 48 8b 93 f0 01 00 00 48 03 14 c5 20 70 67 81 <48> ff 0a 83 3b 02 74 23 bf 01 00 00 00 e8 6d d4 fc ff 65 48 8b [ 1509.674680] RIP [<ffffffff810710f1>] module_put+0x31/0x80 [ 1509.674689] RSP <ffff880119cebf38> [ 1509.674695] ---[ end trace 6aaa73f49bc73faf ]--- [ 1509.674701] note: modprobe[29571] exited with preempt_count 1 after i tried to unload drivers(dont remember nvidia ones or ath9k ones) machine became unresponsive, i had to raise elephants. after which new kernel did not reboot properly. recompiled, it from working kernel, relaoded machine, everything seesm to be working now however, i still get same issue. pwd /home/dimko/compat-wireless-2009-06-20 tcp dump -i wlan0 [start] ip link set wlan0 start iwconfig wlan0 essid dlink dmesg 761.433090] wlan0: authenticated [ 761.433094] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 761.633264] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 761.640078] wlan0: deauthenticated (Reason: 9) [ 762.640264] wlan0: direct probe to AP 00:1e:58:b4:f6:83 try 1 [ 762.643782] wlan0 direct probe responded [ 762.643787] wlan0: authenticate with AP 00:1e:58:b4:f6:83 [ 762.648350] wlan0: authenticated [ 762.648354] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 762.848263] wlan0: associate with AP 00:1e:58:b4:f6:83 [ 762.855024] wlan0: deauthenticated (Reason: 9) i tcpdum [stop] tcpdump -i wlan0 tcpdump: WARNING: wlan0: no IPv4 address assigned tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on wlan0, link-type EN10MB (Ethernet), capture size 96 bytes ^C 0 packets captured 0 packets received by filter 0 packets dropped by kernel Same problem persist... pstree -p init(1)-+-acpid(4077) |-agetty(4579) |-agetty(4580) |-agetty(4581) |-agetty(4582) |-bonobo-activati(5073)---{bonobo-activati}(5074) |-console-kit-dae(4607)-+-{console-kit-dae}(4611) | |-{console-kit-dae}(4615) | |-{console-kit-dae}(4616) | |-{console-kit-dae}(4617) | |-{console-kit-dae}(4618) | |-{console-kit-dae}(4620) | |-{console-kit-dae}(4621) | |-{console-kit-dae}(4622) | |-{console-kit-dae}(4623) | |-{console-kit-dae}(4624) | |-{console-kit-dae}(4625) | |-{console-kit-dae}(4626) | |-{console-kit-dae}(4627) | |-{console-kit-dae}(4628) | |-{console-kit-dae}(4629) | |-{console-kit-dae}(4630) | |-{console-kit-dae}(4631) | |-{console-kit-dae}(4632) | |-{console-kit-dae}(4633) | |-{console-kit-dae}(4634) | |-{console-kit-dae}(4635) | |-{console-kit-dae}(4636) | |-{console-kit-dae}(4637) | |-{console-kit-dae}(4638) | |-{console-kit-dae}(4639) | |-{console-kit-dae}(4640) | |-{console-kit-dae}(4641) | |-{console-kit-dae}(4642) | |-{console-kit-dae}(4643) | |-{console-kit-dae}(4644) | |-{console-kit-dae}(4645) | |-{console-kit-dae}(4646) | |-{console-kit-dae}(4647) | |-{console-kit-dae}(4648) | |-{console-kit-dae}(4649) | |-{console-kit-dae}(4650) | |-{console-kit-dae}(4651) | |-{console-kit-dae}(4652) | |-{console-kit-dae}(4653) | |-{console-kit-dae}(4654) | |-{console-kit-dae}(4655) | |-{console-kit-dae}(4656) | |-{console-kit-dae}(4657) | |-{console-kit-dae}(4658) | |-{console-kit-dae}(4659) | |-{console-kit-dae}(4660) | |-{console-kit-dae}(4661) | |-{console-kit-dae}(4662) | |-{console-kit-dae}(4663) | |-{console-kit-dae}(4664) | |-{console-kit-dae}(4665) | |-{console-kit-dae}(4666) | |-{console-kit-dae}(4667) | |-{console-kit-dae}(4668) | |-{console-kit-dae}(4669) | |-{console-kit-dae}(4670) | |-{console-kit-dae}(4671) | |-{console-kit-dae}(4672) | |-{console-kit-dae}(4673) | |-{console-kit-dae}(4877) | |-{console-kit-dae}(4926) | `-{console-kit-dae}(4983) |-dbus-daemon(4233) |-dbus-daemon(5021) |-dbus-launch(5020) |-dhcpcd(3192) |-exaile(5149) |-gconfd-2(5027) |-gdm(4979)---gdm(4980)-+-X(4982) | `-gnome-session(5005)-+-gnome-panel(5070) | |-metacity(5066) | |-nautilus(5077) | |-seahorse-agent(5035) | |-ssh-agent(5025) | |-vino-server(5084) | `-{gnome-session}(5047) |-gnome-dictionar(5098) |-gnome-keyring-d(5044) |-gnome-power-man(5103) |-gnome-screensav(5065) |-gnome-settings-(5048)---{gnome-settings-}(5052) |-gnome-terminal(5163)-+-bash(5166)---su(5172)---bash(5175) | |-bash(5218)---su(5223)---bash(5226)---pstree(52+ | |-gnome-pty-helpe(5165) | `-{gnome-terminal}(5168) |-gnome-volume-ma(5104) |-gvfs-hal-volume(5102) |-gvfsd(5037) |-gvfsd-burn(5119) |-gvfsd-trash(5107) |-hald(4292)---hald-runner(4293)-+-hald-addon-acpi(4307) | |-hald-addon-cpuf(4306) | |-hald-addon-inpu(4298) | `-hald-addon-stor(4317) |-login(4577)---bash(4878) |-login(4578)---bash(4921) |-mdadm(2738) |-mixer_applet2(5090)---{mixer_applet2}(5117) |-mozilla-launche(5121)---thunderbird-bin(5131)-+-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(513+ | |-{thunderbird-bin}(514+ | |-{thunderbird-bin}(514+ | `-{thunderbird-bin}(514+ |-multiload-apple(5080) |-smartd(4507) |-syslog-ng(4018) `-udevd(932) lsmod Module Size Used by ath9k 245620 0 mac80211 188280 1 ath9k ath 9512 1 ath9k cfg80211 98080 3 ath9k,mac80211,ath rfkill_backport 21520 1 cfg80211 nvidia 8121160 36 f71882fg 28632 0 snd_seq 62016 0 output 3432 0 ext4 241640 1 jbd2 63000 1 ext4 crc16 1960 1 ext4 snd_virtuoso 14564 1 snd_oxygen_lib 34344 1 snd_virtuoso snd_pcm 88816 1 snd_oxygen_lib snd_timer 24536 2 snd_seq,snd_pcm snd_page_alloc 9976 1 snd_pcm snd_mpu401_uart 8424 1 snd_oxygen_lib snd_rawmidi 23336 1 snd_mpu401_uart snd_seq_device 7772 2 snd_seq,snd_rawmidi snd 60968 10 snd_seq,snd_virtuoso,snd_oxygen_lib,snd_pcm,snd_timer,snd_mpu401_uart,snd_rawmidi,snd_seq_device r8169 37100 0 k8temp 5608 0 hwmon 2920 2 f71882fg,k8temp soundcore 1648 1 snd i2c_nforce2 8336 0 i2c_core 27328 2 nvidia,i2c_nforce2

2 1

[alsa-devel] alc889
by Stefan Krastanov 22 Jun '09

22 Jun '09

Hi I would like to help (or at least help myself) with the creation of a model config for alc889. I'm not a sound developer. I'm using a laptop with alc889 based sound card. The reported pins are really messed up (it's hard not to use more profane expression). My laptop is Fujitsu-Siemens Amilo xi but the problems are the same as here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/71205 or here: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-October/011295.ht… namely: 1.- no jack sensing (extremely annoying - for two years allready) 2.- misnamed mixers in volume control for gnome or alsamixer (example: front instead of headphone) 3.- the volume-control potentiometer on the side of the laptop is not working 4.- I need to mute two different channel at the same time (center and LFE) to mute one physical output (build-in sub-woofer) 5.- A great deal of other small annoyances destroying the user experience I'm using model 3stack-dig and the optical output is working (at least is lighted up - I do not have the hardware to check it) My questions are: - What I need to do, to create new model option based on 3stack-dig with different pin mapping? - Is it possible that my present configuration (3stack-dig) is using pins that should not be used? I'm asking in search of an explanation of problem 4. - What I need to do to rename the controls in alsamixer? - What to do to repair jack sensing? - that's the most important of all - What to do to repair the potentiometer for volume control? I have seen that hda-analyser can create diff files, but where should they be used. (And why is hda-analyser not working - I was forced to use http://ubuntuforums.org/showpost.php?p=7227626&postcount=33. Is it that much to expect working tools on the official website. I understand that small bug will always exist, but that one is there for at least 10 months) I won't be surprised if many of my questions are answered in the documentation, but I was unable to find anything useful. It would be enough to point me a place to start reading. I know you guys are working hard and you have done great things, but it's really hard to find something in the documentation(which is occasionally quite old) Best regards and thank you for the great work Stefan Krastanov

2 1

[alsa-devel] [PATCH] sound: remove redundant tests on unsigned
by Roel Kluin 22 Jun '09

22 Jun '09

reg and channel_id are unsigned. When negative they are wrapped and caught by the other test. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 3 +-- sound/pci/emu10k1/emuproc.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- 4 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d1..2deac54 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", &reg, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if ((reg < 0x40) && (val <= 0xffffffff)) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 4d3ad79..957b130 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if ((reg < 0x49) && (val <= 0xffffffff) && (channel_id <= 2)) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f974..6ad3825 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if ((reg < 0xa0) && (val <= 0xffffffff) && (channel_id <= 3)) snd_ptr_write(emu, iobase, reg, channel_id, val); } } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 0b8dcb5..6f8cd8d 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, * of data into val */ - if ((reg < 0 || reg > 9) && (reg != 15)) { + if ((reg > 9) && (reg != 15)) { printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; }

2 1

Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Mark Brown 22 Jun '09

22 Jun '09

On Fri, Jun 19, 2009 at 05:28:15PM +0800, Barry Song wrote: > 1. add AD1938 codec driver (codec) > 2. add blackfin SPORT-TDM DAI and PCM driver (platform) > 3. add bf5xx board with AD1938 driver (machine) As Liam said you really need to submit this as a patch series rather than as a single big patch - as your commit log here indicates you've got several different things going on here. > +++ b/include/sound/soc-dai.h > @@ -30,6 +30,7 @@ struct snd_pcm_substream; > #define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ > #define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ > #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ > +#define SND_SOC_DAIFMT_SPORT_TDM 6 /* SPORT TDM for ADI parts */ If you're going to add a new DAI format that really needs more explanation than this explaining what the DAI format is. It'd be very surprising to see hardware needing a new format. Looking at the datasheet for the ad1938 it appears that the actual format here is just normal I2S with TDM. This does not need a new DAI format or new CPU DAI, you just need to add suport for TDM to the Blackfin I2S driver. The format is fairly standard and implemented by a number of other devices. See set_tdm_slot() for setting up the higher channel counts - there's some ongoing revisions to that API so you'll want to also ensure that the code is set up so that it can cope with specification of the sample width for each slot in set_tdm_slot(). Given this I've only looked at the CODEC driver below. > diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c > new file mode 100644 > index 0000000..9aa78e1 > --- /dev/null > +++ b/sound/soc/codecs/ad1938.c > + * > + * Revision history > + * 4 June 2009 Initial version. Don't include this, git provides code history for us. > +struct snd_soc_device *ad1938_socdev; > + > +/* dac de-emphasis enum control */ > +static const char *ad1938_deemp[] = {"flat", "48kHz", "44.1kHz", "32kHz"}; For consistency with other drivers "flat" should be "None". > +/* AD1938 volume/mute/de-emphasis etc. controls */ > +static const struct snd_kcontrol_new ad1938_snd_controls[] = { > + /* DAC volume control */ > + SOC_SINGLE("DAC L1 Volume", AD1938_DAC_L1_VOL, 0, 0xFF, 1), > + SOC_SINGLE("DAC R1 Volume", AD1938_DAC_R1_VOL, 0, 0xFF, 1), These (and the other stereo pairs below) should be SOC_DOUBLE_R(). This allows ALSA to represent them as stereo controls to applications rather than as two separate controls. You should also provide TLV information so actually SOC_DOUBLE_R_TLV() if possible. > + /* DAC mute control */ > + SOC_SINGLE("DAC L1 Switch", AD1938_DAC_CHNL_MUTE, 0, 1, 1), > + SOC_SINGLE("DAC R1 Switch", AD1938_DAC_CHNL_MUTE, 1, 1, 1), These should be stereo controls too - SOC_DOUBLE() since they're in the same register. > + /* ADC mute control */ > + SOC_SINGLE("ADC L1 Switch", AD1938_ADC_CTRL0, ADC0_MUTE, 1, 1), > + SOC_SINGLE("ADC R1 Switch", AD1938_ADC_CTRL0, ADC1_MUTE, 1, 1), These too. > + /* DAC de-emphasis */ > + SOC_ENUM("Playback Deemphasis", ad1938_enum[0]), Don't put your enums in an array, use named variables for them. This makes drivers easier to maintian when you get a lot of enums. > +static int ad1938_add_controls(struct snd_soc_codec *codec) > +{ > + int err, i; > + > + for (i = 0; i < ARRAY_SIZE(ad1938_snd_controls); i++) { > + err = snd_ctl_add(codec->card, > + snd_soc_cnew(&ad1938_snd_controls[i], codec, NULL)); Use snd_soc_add_controls() here - you can replace the entire function with a call to that. > +/* dac/adc/pll poweron/off functions */ > +static int ad1938_dac_powerctrl(struct snd_soc_codec *codec, int cmd) > +{ > + int reg; > + > + reg = codec->read(codec, AD1938_DAC_CTRL0); > + if (cmd) > + reg &= ~DAC_POWERDOWN; > + else > + reg |= DAC_POWERDOWN; > + codec->write(codec, AD1938_DAC_CTRL0, reg); This should be handled by DAPM - either have a single DAC widget representing all the channels (since you don't appear to have independant control anyway) or have a bunch of dummy DAC widgets and a supply widget representing the actual power control. The same thing applies to the ADCs. > +static int ad1938_set_pll(struct snd_soc_dai *codec_dai, > + int pll_id, unsigned int freq_in, unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + > + if (freq_out) > + ad1938_pll_powerctrl(codec, 1); > + else { > + /* playing while recording, framework will poweroff-poweron pll redundantly */ > + if ((!codec_dai->capture.active) && (!codec_dai->playback.active)) > + ad1938_pll_powerctrl(codec, 0); > + } Hrm. This appears to completely ignore the frequencies supplied for the PLL and just provide power control. I suspect that you can just handle the PLL as a SND_SOC_DAPM_SUPPLY(), there seems to be no need to expose the set_pll() operation and make machine drivers call it given that there isn't any frequency configuration going on. > +static int ad1938_mute(struct snd_soc_dai *dai, int mute) > +{ > + struct snd_soc_codec *codec = dai->codec; > + > + if (!mute) > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0); > + else > + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0xff); > + > + return 0; > +} This isn't going to play well with the explicit mute controls you've got above - it's writing to the same register bits without any coordination. One or the other set of controls ought to be removed. > +static int ad1938_tdm_set(struct snd_soc_codec *codec) > +{ > + codec->write(codec, AD1938_DAC_CTRL0, (codec->read(codec, AD1938_DAC_CTRL0) & > + (~DAC_SERFMT_MASK)) | DAC_SERFMT_TDM); > + codec->write(codec, AD1938_DAC_CTRL1, 0x84); /* invert bclk, 256bclk/frame, latch in mid */ > + codec->write(codec, AD1938_ADC_CTRL1, 0x43); /* sata delay=1, adc aux mode */ > + codec->write(codec, AD1938_ADC_CTRL2, 0x6F); /* left high, driver on rising edge */ > + > + return 0; > +} If you use set_tdm_slot() then the BCLK/frame ratio will be set by that. Inversion of BCLK (and any other clocks) should be handled by the set_dai_fmt() operation based on the machine driver request rather than done unconditionally. > + /* bit size */ > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S16_LE: > + word_len = 3; > + break; Once you implement set_tdm_slot() you should allow the word length to be configured there if it's called or otherwise keep this code here - see Daniel Ribeiro's patche "change set_tdm_slot api to allow slot_width override" posted to the ALSA list this week. > +static int __devinit ad1938_spi_probe(struct spi_device *spi) > +{ > + spi->dev.power.power_state = PMSG_ON; > + ad1938_socdev->card->codec->control_data = spi; > + > + return 0; > +} > + > +static int __devexit ad1938_spi_remove(struct spi_device *spi) > +{ > + return 0; > +} Your device probing should all be restructured so that the SPI device for the CODEC is registered as any other SPI device rather than being set up as part of probing the ASoC device. See the wm8731 driver for an example of doing this for a SPI device. This will require that the arch code for any systems with the ad1938 do the setup of the device. > + .name = "AD1938", > + .playback = { > + .stream_name = "Playback", > + .channels_min = 2, > + .channels_max = 8, > + .rates = SNDRV_PCM_RATE_48000, > + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, Please keep your lines to under 80 columns. > +#define AD1938_PLL_CLK_CTRL0 0 > +#define PLL_POWERDOWN 0x01 > +#define AD1938_PLL_CLK_CTRL1 1 > +#define AD1938_DAC_CTRL0 2 > +#define DAC_POWERDOWN 0x01 > +#define DAC_SERFMT_MASK 0xC0 > +#define DAC_SERFMT_STEREO (0 << 6) > +#define DAC_SERFMT_TDM (1 << 6) These defines need namespacing if they're going to appear in the headers - everything should have the AD1938_ prefix.

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[PATCH] ASoC: Add a shutdown callback
by Mark Brown 22 Jun '09

22 Jun '09

Ensure that the audio subsystem is powered down cleanly when the system shuts down by providing a shutdown operation. This ensures that all the components have been returned to an off state cleanly which should avoid audio issues from partially charged capacitors or noise on digital inputs if the system is restarted quickly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 29 +++++++++++++++++++++++++++++ 3 files changed, 46 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f..35814ce 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4414117..55d45c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1020,6 +1020,21 @@ static int soc_remove(struct platform_device *pdev) return 0; } +static void soc_shutdown(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + + if (!card->instantiated) + return; + + /* Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. */ + run_delayed_work(&card->delayed_work); + + snd_soc_dapm_shutdown(socdev); +} + /* ASoC platform driver */ static struct platform_driver soc_driver = { .driver = { @@ -1030,6 +1045,7 @@ static struct platform_driver soc_driver = { .remove = soc_remove, .suspend = soc_suspend, .resume = soc_resume, + .shutdown = soc_shutdown, }; /* create a new pcm */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6534359..b9129ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2032,6 +2032,35 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_dapm_widget *w; + LIST_HEAD(down_list); + int powerdown = 0; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (w->power) { + dapm_seq_insert(w, &down_list, dapm_down_seq); + powerdown = 1; + } + } + + /* If there were no widgets to power down we're already in + * standby. + */ + if (powerdown) { + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + dapm_seq_run(codec, &down_list, 0, dapm_down_seq); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + } + + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); +} + /* Module information */ MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); -- 1.6.3.1

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