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

[alsa-devel] [PATCH RFC 0/4] Driver for RME MADI FX
by Adrian Knoth 04 Feb '13

04 Feb '13

Hi! I've been developing an out-of-tree driver for the RME MADI FX. History is available at https://github.com/adiknoth/madifx but doesn't follow any kernel convention regarding commit isolation, it was basically a backup/scratchpad during development. I don't have too much feedback, yet, but I guess it makes sense to mainline it earlier than later, increase the driver's exposure and fix any problems when the card is in more widespread use. Note that this patch series is essentially RFC right now to gather developer feedback. Cheers Adrian Knoth (4): ALSA: madifx - Makefile entries for RME MADI FX ALSA: Add RME MADI FX Kconfig entries ALSA: madifx - Expose preliminary userspace interface ALSA: madifx - Add support for RME MADI FX include/uapi/sound/madifx.h | 173 ++ sound/pci/Kconfig | 21 + sound/pci/rme9652/Makefile | 2 + sound/pci/rme9652/madifx.c | 3986 +++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 4182 insertions(+) create mode 100644 include/uapi/sound/madifx.h create mode 100644 sound/pci/rme9652/madifx.c -- 1.7.10.4

2 7

[alsa-devel] [PATCH] ALSA: core: don't return uninitialized snd_compr_tstamp
by Mark Brown 04 Feb '13

04 Feb '13

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> The snd_compr_update_tstamp() can only fill in the snd_compr_tstamp if the codec implements the pointer() function. If that happened the code was previously returning uninitialized garbage in the tstamp because it wasn't initialized anywhere. This change zero-fills the tstamp in the two places it is used before calling snd_compr_update_tstamp(), and also has snd_compr_update_tstamp() return an error indication if it can't provide a tstamp. For the case of snd_compr_calc_avail() it ignores this error because we still need to return info on the available buffer space even if we can't provide tstamp info - when the tstamp is not valid all fields are now guaranteed to be zero. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ad11dc9..2d62068 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -144,16 +144,17 @@ static int snd_compr_free(struct inode *inode, struct file *f) return 0; } -static void snd_compr_update_tstamp(struct snd_compr_stream *stream, +static int snd_compr_update_tstamp(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { if (!stream->ops->pointer) - return; + return -ENOTSUPP; stream->ops->pointer(stream, tstamp); pr_debug("dsp consumed till %d total %d bytes\n", tstamp->byte_offset, tstamp->copied_total); stream->runtime->hw_pointer = tstamp->byte_offset; stream->runtime->total_bytes_transferred = tstamp->copied_total; + return 0; } static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, @@ -161,7 +162,9 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, { long avail_calc; /*this needs to be signed variable */ + memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); + /* Still need to return avail even if tstamp can't be filled in */ /* FIXME: This needs to be different for capture stream, available is # of compressed data, for playback it's @@ -517,11 +520,14 @@ out: static inline int snd_compr_tstamp(struct snd_compr_stream *stream, unsigned long arg) { - struct snd_compr_tstamp tstamp; + struct snd_compr_tstamp tstamp = {0}; + int ret; - snd_compr_update_tstamp(stream, &tstamp); - return copy_to_user((struct snd_compr_tstamp __user *)arg, - &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + ret = snd_compr_update_tstamp(stream, &tstamp); + if (ret == 0) + ret = copy_to_user((struct snd_compr_tstamp __user *)arg, + &tstamp, sizeof(tstamp)) ? -EFAULT : 0; + return ret; } static int snd_compr_pause(struct snd_compr_stream *stream) -- 1.7.10.4

4 3

[alsa-devel] [PATCH] ASOC: imx: audmux: add a integrated configuration function
by Gary Zhang 04 Feb '13

04 Feb '13

based on imx_audmux_v2_configure_port(), add a integrated interface to configure audmux port conveniently Signed-off-by: Gary Zhang <b13634(a)freescale.com> --- sound/soc/fsl/imx-audmux.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 42 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d9..ebd38a0 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -1,5 +1,5 @@ /* - * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012-2013 Freescale Semiconductor, Inc. * Copyright 2012 Linaro Ltd. * Copyright 2009 Pengutronix, Sascha Hauer <s.hauer(a)pengutronix.de> * @@ -244,6 +244,47 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); +int imx_audmux_configure_port(unsigned int int_port, unsigned int ext_port) +{ + int ret; + + if ((int_port > 7) || (ext_port > 7)) + return -EINVAL; + + if (audmux_type != IMX31_AUDMUX) + return -EINVAL; + + if (!audmux_base) + return -ENOSYS; + + if (audmux_clk) + clk_prepare_enable(audmux_clk); + + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + pr_err("audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + pr_err("audmux external port setup failed\n"); + return ret; + } + + return 0; +} +EXPORT_SYMBOL_GPL(imx_audmux_configure_port); + static int imx_audmux_probe(struct platform_device *pdev) { struct resource *res; -- 1.7.0.4

3 5

[alsa-devel] [PATCH] ASoC: wm2200: Add Rx ANC input select control
by Chris Rattray 04 Feb '13

04 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Change-Id: Iee28d1898327633a3166dd340410a37796959c0c --- sound/soc/codecs/wm2200.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index bdd3d80..a915c76 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1109,6 +1109,16 @@ static int wm2200_mixer_values[] = { static WM2200_MUX_CTL_DECL(name##_aux5); \ static WM2200_MUX_CTL_DECL(name##_aux6); +static const char *wm2200_rxanc_input_sel_texts[] = { + "None", "IN1", "IN2", "IN3", +}; + +static const struct soc_enum wm2200_rxanc_input_sel = + SOC_ENUM_SINGLE(WM2200_RXANC_SRC, + WM2200_IN_RXANC_SEL_SHIFT, + ARRAY_SIZE(wm2200_rxanc_input_sel_texts), + wm2200_rxanc_input_sel_texts); + static const struct snd_kcontrol_new wm2200_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL, WM2200_IN1_OSR_SHIFT, 1, 0), @@ -1168,6 +1178,7 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), +SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm2200: correct IN2L and IN3L digital mute
by Chris Rattray 04 Feb '13

04 Feb '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index a915c76..f5410ec 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1136,9 +1136,9 @@ SOC_DOUBLE_R_TLV("IN3 Volume", WM2200_IN3L_CONTROL, WM2200_IN3R_CONTROL, SOC_DOUBLE_R("IN1 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, +SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_2L, WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, +SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_3L, WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_1L, -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix default multichannel HDMI mapping regression
by Anssi Hannula 04 Feb '13

04 Feb '13

Commit d45e6889ee69456a4d5b1bbb32252f460cd48fa9 ("ALSA: hda - Provide the proper channel mapping for generic HDMI driver") added support for custom channel maps in the HDA HDMI driver. Due to a mistake in an 'if' condition the custom map is always used even when no such map has been set. This causes incorrect channel mapping for multichannel audio by default. Pass per_pin->chmap_set to hdmi_setup_channel_mapping() as a parameter so that it can use it for detecting if a custom map has been set instead of checking if map is NULL (which is never the case). Reported-by: Staffan Lindberg <pike(a)xbmc.org> Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: stable(a)vger.kernel.org --- This also affects stable 3.7, but not earlier versions. sound/pci/hda/patch_hdmi.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 807a2aa..e85959f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -714,9 +714,10 @@ static void hdmi_setup_fake_chmap(unsigned char *map, int ca) static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, bool non_pcm, int ca, - int channels, unsigned char *map) + int channels, unsigned char *map, + bool chmap_set) { - if (!non_pcm && map) { + if (!non_pcm && chmap_set) { hdmi_manual_setup_channel_mapping(codec, pin_nid, channels, map); } else { @@ -905,7 +906,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, pin_nid, channels); hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap); + channels, per_pin->chmap, + per_pin->chmap_set); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); @@ -915,7 +917,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * accordingly */ if (per_pin->non_pcm != non_pcm) hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap); + channels, per_pin->chmap, + per_pin->chmap_set); } per_pin->non_pcm = non_pcm; -- 1.7.10

2 1

[alsa-devel] Loopback interface: suspend/resume leads to a kernel crash
by El boulangero 04 Feb '13

04 Feb '13

Hi, I use the loopback device all the time. It works pretty good, except when I suspend my computer to RAM. On resume, I've got a kernel call trace. Here is the procedure to reproduce this behaviour, it's pretty easy. # Insert module if needed $ modprobe snd_aloop pcm_substreams=2 # Let's see how it looks like $ aplay -l **** Liste des Périphériques Matériels PLAYBACK **** carte 0: PCH [HDA Intel PCH], périphérique 0: VT1802 Analog [VT1802 Analog] Sous-périphériques: 1/1 Sous-périphérique #0: subdevice #0 carte 0: PCH [HDA Intel PCH], périphérique 2: VT1802 HP [VT1802 HP] Sous-périphériques: 1/1 Sous-périphérique #0: subdevice #0 carte 0: PCH [HDA Intel PCH], périphérique 3: HDMI 0 [HDMI 0] Sous-périphériques: 1/1 Sous-périphérique #0: subdevice #0 carte 1: Loopback [Loopback], périphérique 0: Loopback PCM [Loopback PCM] Sous-périphériques: 2/2 Sous-périphérique #0: subdevice #0 Sous-périphérique #1: subdevice #1 carte 1: Loopback [Loopback], périphérique 1: Loopback PCM [Loopback PCM] Sous-périphériques: 2/2 Sous-périphérique #0: subdevice #0 Sous-périphérique #1: subdevice #1 # Use the loopback device $ aplay -D hw:1,0,0 sound.wave & # Suspend computer $ pm-suspend # Wake up the computer That's enough to reproduce this bug. Here are the kernel logs: [ 181.994434] PM: resume of devices complete after 3301.753 msecs [ 182.006413] Restarting tasks ... done. [ 182.006590] ------------[ cut here ]------------ [ 182.006605] kernel BUG at kernel/timer.c:908! [ 182.006619] video LNXVIDEO:00: Restoring backlight state [ 182.006621] invalid opcode: 0000 [#1] PREEMPT SMP [ 182.006640] Modules linked in: snd_aloop af_packet nfsd auth_rpcgss nfs_acl nfs lockd fscache sunrpc ipv6 loop uvcvideo videobuf2_vmalloc videobuf2_memops videobuf2_core videodev media joydev hid_generic usbhid coretemp kvm_intel kvm crc32c_intel ghash_clmulni_intel aesni_intel xts aes_x86_64 lrw i915 gf128mul i2c_algo_bit intel_agp intel_gtt drm_kms_helper ablk_helper drm cryptd iTCO_wdt snd_hda_codec_hdmi i2c_i801 psmouse iTCO_vendor_support snd_hda_codec_via snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_page_alloc iwlwifi cfg80211 snd_seq snd_seq_device snd_timer snd ehci_pci button battery ac rtc_cmos microcode video i2c_core soundcore lpc_ich evdev serio_raw mei wmi ehci_hcd rfkill ext4 crc16 jbd2 mbcache sg sr_mod sd_mod cdrom ahci libahci libata scsi_mod thermal r8169 mii xhci_hcd [ 182.006642] CPU 1 [ 182.006642] Pid: 2982, comm: aplay Not tainted 3.8.0-rc5-kdump #1 System76, Inc. Gazelle Professional/Gazelle Professional [ 182.006646] RIP: 0010:[<ffffffff81063f8f>] [<ffffffff81063f8f>] add_timer+0xf/0x20 [ 182.006647] RSP: 0018:ffff880214a93d40 EFLAGS: 00010082 [ 182.006647] RAX: 00000000fffe34c9 RBX: ffff88021570e200 RCX: 00000000fffe344c [ 182.006648] RDX: 000000000001f3ff RSI: 0000000000f42400 RDI: ffff88021492f350 [ 182.006648] RBP: ffff88021492f300 R08: 000000000001f400 R09: 0000000000000020 [ 182.006649] R10: ffff88021588f800 R11: 0000000000000206 R12: 00000000fffffffe [ 182.006649] R13: ffff880214a3cc00 R14: ffff880214bbc464 R15: ffff88021588f800 [ 182.006650] FS: 00007f3489ed0700(0000) GS:ffff88021f240000(0000) knlGS:0000000000000000 [ 182.006650] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 182.006651] CR2: 0000000000000000 CR3: 00000002148fe000 CR4: 00000000001407e0 [ 182.006652] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 182.006652] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 182.006653] Process aplay (pid: 2982, threadinfo ffff880214a92000, task ffff880215928000) [ 182.006653] Stack: [ 182.006654] ffffffffa0d0947c 00000000000000b4 00000b2c00000003 0000000000003e80 [ 182.006655] ffffffffa02fbe80 ffff880214a3cc00 0000000000000003 ffff880214a3cc00 [ 182.006656] 0000000000000000 0000000000000fa0 ffffffffa02f01b0 0000000000003e80 [ 182.006656] Call Trace: [ 182.006659] [<ffffffffa0d0947c>] ? loopback_trigger+0x20c/0x2e0 [snd_aloop] [ 182.006664] [<ffffffffa02f01b0>] ? snd_pcm_action_single+0x30/0x80 [snd_pcm] [ 182.006667] [<ffffffffa02f86d1>] ? snd_pcm_lib_write1+0x391/0x400 [snd_pcm] [ 182.006669] [<ffffffffa02f5ab0>] ? snd_pcm_lib_readv_transfer+0x170/0x170 [snd_pcm] [ 182.006672] [<ffffffffa02f3f05>] ? snd_pcm_playback_ioctl1+0x255/0x270 [snd_pcm] [ 182.006674] [<ffffffffa02f3f47>] ? snd_pcm_playback_ioctl+0x27/0x30 [snd_pcm] [ 182.006676] [<ffffffff8115f43f>] ? do_vfs_ioctl+0x8f/0x530 [ 182.006678] [<ffffffff8114e4de>] ? vfs_read+0xfe/0x180 [ 182.006679] [<ffffffff8115f980>] ? sys_ioctl+0xa0/0xc0 [ 182.006681] [<ffffffff814591ad>] ? system_call_fastpath+0x1a/0x1f [ 182.006691] Code: e9 4c fe ff ff 4d 89 f4 e9 d4 fe ff ff 66 66 66 66 66 66 2e 0f 1f 84 00 00 00 00 00 48 83 3f 00 75 09 48 8b 77 10 e9 a1 fd ff ff <0f> 0b 66 66 66 66 66 66 2e 0f 1f 84 00 00 00 00 00 48 83 ec 08 [ 182.006692] RIP [<ffffffff81063f8f>] add_timer+0xf/0x20 [ 182.006692] RSP <ffff880214a93d40> I tried to change a few things in the snd_aloop driver, but this code is way beyond my understanding. I can't fix that myself. Feel free to contact me if you need more information. Thanks ! Regards, Arnaud

2 4

[alsa-devel] [PATCH] UAC2 clock selector - fixed?
by Louis Gorenfeld 03 Feb '13

03 Feb '13

Hi, I'd been struggling with a UAC2 device that I couldn't set the clock on. I finally got in touch with a driver author who pointed me to some official documentation (the USB Device Class Definition for Audio Devices). Where the ALSA code always sends a 16-bit packet to control mixer settings, the docs say that the clock selector control uses layout 1 which is 8-bit. By changing the routine to send an 8-bit packet, that control now works. This is the patch that fixes the problem for me (for ALSA 1.0.25). Is this proper? --- a/linux-3.4/sound/usb/mixer.c 2012-05-20 15:29:13.000000000 -0700 +++ b/vanilla-3.4/sound/usb/mixer.c 2013-01-02 17:06:11.009660059 -0800 @@ -422,8 +422,13 @@ if (cval->mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + /* audio class v2 controls are often but not always 2 bytes in size */ + /* For example, the clock selector (see 5.2.5.2.1 in the Universal Serial + * Bus Device Class Definition for Audio Devices document) */ + if (cval->control == UAC2_CX_CLOCK_SELECTOR) + val_len = sizeof(__u8); + else + val_len = sizeof(__u16); /* FIXME */ if (request != UAC_SET_CUR) { Thanks much! Louis

2 1

[alsa-devel] LAC 2013: the Linux Audio Conference - Deadline Extension
by IOhannes m zmoelnig - LAC 2013 Organizing Team 02 Feb '13

02 Feb '13

sorry for >< please >> << Hi all, The Linux Audio Conference submissions deadline has been extended! It is now February 17th, 2013 (23:59 HAST) So, if you were considering to submit a paper but couldn't make up your mind yet, here is your chance to become active! Never forget that this conference lives through the people participating in it. FEBRUARY 17th is the new deadline for all submission types: papers, music, installations, workshop proposals. Check out the link below more info: http://lac.linuxaudio.org/2013/participation Please spread this information to anyone who might be interested. If you have any questions, drop us a line at lac(a)linuxaudio.org We are looking forward to seeing you in Graz in May! Thanks and happy last-minute music-and-paper-submissions, on behalf of the LAC2013 organization team, IOhannes m zmoelnig --- LAC 2013: the Linux Audio Conference May 9-12, 2013 @ IEM, Graz/Austria lac(a)linuxaudio.org http://lac.iem.at

1 0

[alsa-devel] Lenovo ThinkCentre A58 - surround 5.1 problem
by Raymond Yau 02 Feb '13

02 Feb '13

> I already tried so many options in alsa-mixer - as far as they where possible to set. As we say im Germany: "I'm out of my latin" ;-) > > Any Ideas how to go on? > >> > >> > I'm running a Lenovo ThinkCentre A58 with a ALC662 rev. 1 chip on >> board. The mainboard has 3 jacks at the back of the PC and 2 ones at >> it's front (mic and headphones). >> > >> > The problem is, that the connectors are not automatically arranged to >> playback 5.1 surround sound. I must manually set the "model" to >> "asusmode4" to get 5.1 and then run a script as root, to set the >> backside jacks to output and to Center/LFE and rear speaker. Also I have >> to mute the internal speaker. >> > >> > I've tried to patch the module manually, but this went wrong somehow. >> I don't know how to proceed. >> > >> > So maybe you can help me and I can help you to complete the ALSA code >> for this special mainboard/PC - hopefully. >> > >> > So long, I would thank you for your feedback. >> > >> For a desktop with/without internal speaker, three audio jacks at rear panel Green lineout jack , blue and pink jacks should have higher priority to allocate DAC (audio output node 0x02, 0x03 and 0x04) than the headphone and the internal speaker to support surround 5.1 This mean that in function parse_output_paths() to prevent the assignment of dac speaker - bool fill_hardwired = true, fill_mio_first = true; + bool fill_hardwired = false, fill_mio_first = true; Or use the existence of internal Mic to determine the value of fill_hardwired

1 0

[alsa-devel] New one on board - a damn rooky :">
by Rüdi 02 Feb '13

02 Feb '13

Hello @ all, I'm Rüdiger from Germany. I'm no programmer, that's my biggest fault here. But because of a "private" problem with ALSA and Intel HDA I want to help whereever I can - at least with that special problem. I'm running a Lenovo ThinkCentre A58 with a ALC662 rev. 1 chip on board. The mainboard has 3 jacks at the back of the PC and 2 ones at it's front (mic and headphones). The problem is, that the connectors are not automatically arranged to playback 5.1 surround sound. I must manually set the "model" to "asusmode4" to get 5.1 and then run a script as root, to set the backside jacks to output and to Center/LFE and rear speaker. Also I have to mute the internal speaker. I've tried to patch the module manually, but this went wrong somehow. I don't know how to proceed. So maybe you can help me and I can help you to complete the ALSA code for this special mainboard/PC - hopefully. So long, I would thank you for your feedback. Regards, Rüdiger

2 2

[alsa-devel] [PATCH - my module 1/1] Fix a memory access violation resulting from improper error propogation.
by allan＠allanwirth.com 01 Feb '13

01 Feb '13

From: Allan Wirth <allan(a)allanwirth.com> Fixes an issue where a variable is used undeclared, which can cause seg faults on some systems if the configuration file is not formatted properly. Signed-off-by: Allan Wirth <allan(a)allanwirth.com> diff --git a/src/conf.c b/src/conf.c index ecaa339..4ba5863 100644 --- a/src/conf.c +++ b/src/conf.c @@ -3530,7 +3530,7 @@ int snd_config_hook_load(snd_config_t *root, snd_config_t *config, snd_config_t if (err < 0) goto _err; } - } else if (config_file_open(root, fi[idx].name) < 0) + } else if ((err = config_file_open(root, fi[idx].name)) < 0) goto _err; } *dst = NULL; -- 1.7.10.4

2 1

[alsa-devel] [git pull] another usb-audio bugfix
by Clemens Ladisch 01 Feb '13

01 Feb '13

The following change since commit 949db153b6466c6f7cad5a427ecea94985927311: Linux 3.8-rc5 (2013-01-25 11:57:28 -0800) is available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git usb-audio-fix for you to fetch changes up to 7da58046482fceb17c4a0d4afefd9507ec56de7f: ALSA: usb-audio: fix Roland A-PRO support (2013-01-31 21:21:59 +0100) ---------------------------------------------------------------- Clemens Ladisch (1): ALSA: usb-audio: fix Roland A-PRO support sound/usb/quirks-table.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) ---------------------------------------------------------------- ALSA: usb-audio: fix Roland A-PRO support The quirk for the Roland/Cakewalk A-PRO keyboards accidentally used the wrong interface number, which prevented the driver from attaching to the device. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> Cc: 2.6.37+ <stable(a)vger.kernel.org> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 64d25a7..820580a 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1750,7 +1750,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "Roland", */ /* .product_name = "A-PRO", */ - .ifnum = 1, + .ifnum = 0, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0003,

2 1

[alsa-devel] [PATCH 002/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 01 Feb '13

01 Feb '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch completes the replacement of the existing max98090 driver, by installing a more complete driver. Signed-off-by: Jerry Wong <jerry.wong(a)maximintegrated.com> Tested-by: Matthew Mowdy <matthew.mowdy(a)maximintegrated.com> Reviewed-by: Ralph Birt <ralph.birt(a)maximintegrated.com> --- Mark, I have rebased and remade the patches. Jerry include/sound/max98090.h | 29 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98090.c | 2398 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98090.h | 1549 ++++++++++++++++++++++++++++ 5 files changed, 3982 insertions(+), 0 deletions(-) create mode 100755 include/sound/max98090.h create mode 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h diff --git a/include/sound/max98090.h b/include/sound/max98090.h new file mode 100755 index 0000000..95efb13 --- /dev/null +++ b/include/sound/max98090.h @@ -0,0 +1,29 @@ +/* + * Platform data for MAX98090 + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SOUND_MAX98090_PDATA_H__ +#define __SOUND_MAX98090_PDATA_H__ + +/* codec platform data */ +struct max98090_pdata { + + /* Analog/digital microphone configuration: + * 0 = analog microphone input (normal setting) + * 1 = digital microphone input + */ + unsigned int digmic_left_mode:1; + unsigned int digmic_right_mode:1; + unsigned int digmic_3_mode:1; + unsigned int digmic_4_mode:1; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0e368d4..3a84782 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -267,6 +268,9 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate +config SND_SOC_MAX98090 + tristate + config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aa56312..f6e8e36 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o +snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -157,6 +158,7 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c new file mode 100755 index 0000000..590ac94 --- /dev/null +++ b/sound/soc/codecs/max98090.c @@ -0,0 +1,2398 @@ +/* + * max98090.c -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/max98090.h> +#include "max98090.h" + +#include <linux/version.h> + +#define DEBUG +#define EXTMIC_METHOD +#define EXTMIC_METHOD_TEST + +/* Allows for sparsely populated register maps */ +static struct reg_default max98090_reg[] = { + { 0x00, 0x00 }, /* 00 Software Reset */ + { 0x03, 0x04 }, /* 03 Interrupt Masks */ + { 0x04, 0x00 }, /* 04 System Clock Quick */ + { 0x05, 0x00 }, /* 05 Sample Rate Quick */ + { 0x06, 0x00 }, /* 06 DAI Interface Quick */ + { 0x07, 0x00 }, /* 07 DAC Path Quick */ + { 0x08, 0x00 }, /* 08 Mic/Direct to ADC Quick */ + { 0x09, 0x00 }, /* 09 Line to ADC Quick */ + { 0x0A, 0x00 }, /* 0A Analog Mic Loop Quick */ + { 0x0B, 0x00 }, /* 0B Analog Line Loop Quick */ + { 0x0C, 0x00 }, /* 0C Reserved */ + { 0x0D, 0x00 }, /* 0D Input Config */ + { 0x0E, 0x1B }, /* 0E Line Input Level */ + { 0x0F, 0x00 }, /* 0F Line Config */ + + { 0x10, 0x14 }, /* 10 Mic1 Input Level */ + { 0x11, 0x14 }, /* 11 Mic2 Input Level */ + { 0x12, 0x00 }, /* 12 Mic Bias Voltage */ + { 0x13, 0x00 }, /* 13 Digital Mic Config */ + { 0x14, 0x00 }, /* 14 Digital Mic Mode */ + { 0x15, 0x00 }, /* 15 Left ADC Mixer */ + { 0x16, 0x00 }, /* 16 Right ADC Mixer */ + { 0x17, 0x03 }, /* 17 Left ADC Level */ + { 0x18, 0x03 }, /* 18 Right ADC Level */ + { 0x19, 0x00 }, /* 19 ADC Biquad Level */ + { 0x1A, 0x00 }, /* 1A ADC Sidetone */ + { 0x1B, 0x00 }, /* 1B System Clock */ + { 0x1C, 0x00 }, /* 1C Clock Mode */ + { 0x1D, 0x00 }, /* 1D Any Clock 1 */ + { 0x1E, 0x00 }, /* 1E Any Clock 2 */ + { 0x1F, 0x00 }, /* 1F Any Clock 3 */ + + { 0x20, 0x00 }, /* 20 Any Clock 4 */ + { 0x21, 0x00 }, /* 21 Master Mode */ + { 0x22, 0x00 }, /* 22 Interface Format */ + { 0x23, 0x00 }, /* 23 TDM Format 1*/ + { 0x24, 0x00 }, /* 24 TDM Format 2*/ + { 0x25, 0x00 }, /* 25 I/O Configuration */ + { 0x26, 0x80 }, /* 26 Filter Config */ + { 0x27, 0x00 }, /* 27 DAI Playback Level */ + { 0x28, 0x00 }, /* 28 EQ Playback Level */ + { 0x29, 0x00 }, /* 29 Left HP Mixer */ + { 0x2A, 0x00 }, /* 2A Right HP Mixer */ + { 0x2B, 0x00 }, /* 2B HP Control */ + { 0x2C, 0x1A }, /* 2C Left HP Volume */ + { 0x2D, 0x1A }, /* 2D Right HP Volume */ + { 0x2E, 0x00 }, /* 2E Left Spk Mixer */ + { 0x2F, 0x00 }, /* 2F Right Spk Mixer */ + + { 0x30, 0x00 }, /* 30 Spk Control */ + { 0x31, 0x2C }, /* 31 Left Spk Volume */ + { 0x32, 0x2C }, /* 32 Right Spk Volume */ + { 0x33, 0x00 }, /* 33 ALC Timing */ + { 0x34, 0x00 }, /* 34 ALC Compressor */ + { 0x35, 0x00 }, /* 35 ALC Expander */ + { 0x36, 0x00 }, /* 36 ALC Gain */ + { 0x37, 0x00 }, /* 37 Rcv/Line OutL Mixer */ + { 0x38, 0x00 }, /* 38 Rcv/Line OutL Control */ + { 0x39, 0x15 }, /* 39 Rcv/Line OutL Volume */ + { 0x3A, 0x00 }, /* 3A Line OutR Mixer */ + { 0x3B, 0x00 }, /* 3B Line OutR Control */ + { 0x3C, 0x15 }, /* 3C Line OutR Volume */ + { 0x3D, 0x00 }, /* 3D Jack Detect */ + { 0x3E, 0x00 }, /* 3E Input Enable */ + { 0x3F, 0x00 }, /* 3F Output Enable */ + + { 0x40, 0x00 }, /* 40 Level Control */ + { 0x41, 0x00 }, /* 41 DSP Filter Enable */ + { 0x42, 0x00 }, /* 42 Bias Control */ + { 0x43, 0x00 }, /* 43 DAC Control */ + { 0x44, 0x06 }, /* 44 ADC Control */ + { 0x45, 0x00 }, /* 45 Device Shutdown */ + { 0x46, 0x00 }, /* 46 Equalizer Band 1 Coefficient B0 */ + { 0x47, 0x00 }, /* 47 Equalizer Band 1 Coefficient B0 */ + { 0x48, 0x00 }, /* 48 Equalizer Band 1 Coefficient B0 */ + { 0x49, 0x00 }, /* 49 Equalizer Band 1 Coefficient B1 */ + { 0x4A, 0x00 }, /* 4A Equalizer Band 1 Coefficient B1 */ + { 0x4B, 0x00 }, /* 4B Equalizer Band 1 Coefficient B1 */ + { 0x4C, 0x00 }, /* 4C Equalizer Band 1 Coefficient B2 */ + { 0x4D, 0x00 }, /* 4D Equalizer Band 1 Coefficient B2 */ + { 0x4E, 0x00 }, /* 4E Equalizer Band 1 Coefficient B2 */ + { 0x4F, 0x00 }, /* 4F Equalizer Band 1 Coefficient A1 */ + + { 0x50, 0x00 }, /* 50 Equalizer Band 1 Coefficient A1 */ + { 0x51, 0x00 }, /* 51 Equalizer Band 1 Coefficient A1 */ + { 0x52, 0x00 }, /* 52 Equalizer Band 1 Coefficient A2 */ + { 0x53, 0x00 }, /* 53 Equalizer Band 1 Coefficient A2 */ + { 0x54, 0x00 }, /* 54 Equalizer Band 1 Coefficient A2 */ + { 0x55, 0x00 }, /* 55 Equalizer Band 2 Coefficient B0 */ + { 0x56, 0x00 }, /* 56 Equalizer Band 2 Coefficient B0 */ + { 0x57, 0x00 }, /* 57 Equalizer Band 2 Coefficient B0 */ + { 0x58, 0x00 }, /* 58 Equalizer Band 2 Coefficient B1 */ + { 0x59, 0x00 }, /* 59 Equalizer Band 2 Coefficient B1 */ + { 0x5A, 0x00 }, /* 5A Equalizer Band 2 Coefficient B1 */ + { 0x5B, 0x00 }, /* 5B Equalizer Band 2 Coefficient B2 */ + { 0x5C, 0x00 }, /* 5C Equalizer Band 2 Coefficient B2 */ + { 0x5D, 0x00 }, /* 5D Equalizer Band 2 Coefficient B2 */ + { 0x5E, 0x00 }, /* 5E Equalizer Band 2 Coefficient A1 */ + { 0x5F, 0x00 }, /* 5F Equalizer Band 2 Coefficient A1 */ + + { 0x60, 0x00 }, /* 60 Equalizer Band 2 Coefficient A1 */ + { 0x61, 0x00 }, /* 61 Equalizer Band 2 Coefficient A2 */ + { 0x62, 0x00 }, /* 62 Equalizer Band 2 Coefficient A2 */ + { 0x63, 0x00 }, /* 63 Equalizer Band 2 Coefficient A2 */ + { 0x64, 0x00 }, /* 64 Equalizer Band 3 Coefficient B0 */ + { 0x65, 0x00 }, /* 65 Equalizer Band 3 Coefficient B0 */ + { 0x66, 0x00 }, /* 66 Equalizer Band 3 Coefficient B0 */ + { 0x67, 0x00 }, /* 67 Equalizer Band 3 Coefficient B1 */ + { 0x68, 0x00 }, /* 68 Equalizer Band 3 Coefficient B1 */ + { 0x69, 0x00 }, /* 69 Equalizer Band 3 Coefficient B1 */ + { 0x6A, 0x00 }, /* 6A Equalizer Band 3 Coefficient B2 */ + { 0x6B, 0x00 }, /* 6B Equalizer Band 3 Coefficient B2 */ + { 0x6C, 0x00 }, /* 6C Equalizer Band 3 Coefficient B2 */ + { 0x6D, 0x00 }, /* 6D Equalizer Band 3 Coefficient A1 */ + { 0x6E, 0x00 }, /* 6E Equalizer Band 3 Coefficient A1 */ + { 0x6F, 0x00 }, /* 6F Equalizer Band 3 Coefficient A1 */ + + { 0x70, 0x00 }, /* 70 Equalizer Band 3 Coefficient A2 */ + { 0x71, 0x00 }, /* 71 Equalizer Band 3 Coefficient A2 */ + { 0x72, 0x00 }, /* 72 Equalizer Band 3 Coefficient A2 */ + { 0x73, 0x00 }, /* 73 Equalizer Band 4 Coefficient B0 */ + { 0x74, 0x00 }, /* 74 Equalizer Band 4 Coefficient B0 */ + { 0x75, 0x00 }, /* 75 Equalizer Band 4 Coefficient B0 */ + { 0x76, 0x00 }, /* 76 Equalizer Band 4 Coefficient B1 */ + { 0x77, 0x00 }, /* 77 Equalizer Band 4 Coefficient B1 */ + { 0x78, 0x00 }, /* 78 Equalizer Band 4 Coefficient B1 */ + { 0x79, 0x00 }, /* 79 Equalizer Band 4 Coefficient B2 */ + { 0x7A, 0x00 }, /* 7A Equalizer Band 4 Coefficient B2 */ + { 0x7B, 0x00 }, /* 7B Equalizer Band 4 Coefficient B2 */ + { 0x7C, 0x00 }, /* 7C Equalizer Band 4 Coefficient A1 */ + { 0x7D, 0x00 }, /* 7D Equalizer Band 4 Coefficient A1 */ + { 0x7E, 0x00 }, /* 7E Equalizer Band 4 Coefficient A1 */ + { 0x7F, 0x00 }, /* 7F Equalizer Band 4 Coefficient A2 */ + + { 0x80, 0x00 }, /* 80 Equalizer Band 4 Coefficient A2 */ + { 0x81, 0x00 }, /* 81 Equalizer Band 4 Coefficient A2 */ + { 0x82, 0x00 }, /* 82 Equalizer Band 5 Coefficient B0 */ + { 0x83, 0x00 }, /* 83 Equalizer Band 5 Coefficient B0 */ + { 0x84, 0x00 }, /* 84 Equalizer Band 5 Coefficient B0 */ + { 0x85, 0x00 }, /* 85 Equalizer Band 5 Coefficient B1 */ + { 0x86, 0x00 }, /* 86 Equalizer Band 5 Coefficient B1 */ + { 0x87, 0x00 }, /* 87 Equalizer Band 5 Coefficient B1 */ + { 0x88, 0x00 }, /* 88 Equalizer Band 5 Coefficient B2 */ + { 0x89, 0x00 }, /* 89 Equalizer Band 5 Coefficient B2 */ + { 0x8A, 0x00 }, /* 8A Equalizer Band 5 Coefficient B2 */ + { 0x8B, 0x00 }, /* 8B Equalizer Band 5 Coefficient A1 */ + { 0x8C, 0x00 }, /* 8C Equalizer Band 5 Coefficient A1 */ + { 0x8D, 0x00 }, /* 8D Equalizer Band 5 Coefficient A1 */ + { 0x8E, 0x00 }, /* 8E Equalizer Band 5 Coefficient A2 */ + { 0x8F, 0x00 }, /* 8F Equalizer Band 5 Coefficient A2 */ + + { 0x90, 0x00 }, /* 90 Equalizer Band 5 Coefficient A2 */ + { 0x91, 0x00 }, /* 91 Equalizer Band 6 Coefficient B0 */ + { 0x92, 0x00 }, /* 92 Equalizer Band 6 Coefficient B0 */ + { 0x93, 0x00 }, /* 93 Equalizer Band 6 Coefficient B0 */ + { 0x94, 0x00 }, /* 94 Equalizer Band 6 Coefficient B1 */ + { 0x95, 0x00 }, /* 95 Equalizer Band 6 Coefficient B1 */ + { 0x96, 0x00 }, /* 96 Equalizer Band 6 Coefficient B1 */ + { 0x97, 0x00 }, /* 97 Equalizer Band 6 Coefficient B2 */ + { 0x98, 0x00 }, /* 98 Equalizer Band 6 Coefficient B2 */ + { 0x99, 0x00 }, /* 99 Equalizer Band 6 Coefficient B2 */ + { 0x9A, 0x00 }, /* 9A Equalizer Band 6 Coefficient A1 */ + { 0x9B, 0x00 }, /* 9B Equalizer Band 6 Coefficient A1 */ + { 0x9C, 0x00 }, /* 9C Equalizer Band 6 Coefficient A1 */ + { 0x9D, 0x00 }, /* 9D Equalizer Band 6 Coefficient A2 */ + { 0x9E, 0x00 }, /* 9E Equalizer Band 6 Coefficient A2 */ + { 0x9F, 0x00 }, /* 9F Equalizer Band 6 Coefficient A2 */ + + { 0xA0, 0x00 }, /* A0 Equalizer Band 7 Coefficient B0 */ + { 0xA1, 0x00 }, /* A1 Equalizer Band 7 Coefficient B0 */ + { 0xA2, 0x00 }, /* A2 Equalizer Band 7 Coefficient B0 */ + { 0xA3, 0x00 }, /* A3 Equalizer Band 7 Coefficient B1 */ + { 0xA4, 0x00 }, /* A4 Equalizer Band 7 Coefficient B1 */ + { 0xA5, 0x00 }, /* A5 Equalizer Band 7 Coefficient B1 */ + { 0xA6, 0x00 }, /* A6 Equalizer Band 7 Coefficient B2 */ + { 0xA7, 0x00 }, /* A7 Equalizer Band 7 Coefficient B2 */ + { 0xA8, 0x00 }, /* A8 Equalizer Band 7 Coefficient B2 */ + { 0xA9, 0x00 }, /* A9 Equalizer Band 7 Coefficient A1 */ + { 0xAA, 0x00 }, /* AA Equalizer Band 7 Coefficient A1 */ + { 0xAB, 0x00 }, /* AB Equalizer Band 7 Coefficient A1 */ + { 0xAC, 0x00 }, /* AC Equalizer Band 7 Coefficient A2 */ + { 0xAD, 0x00 }, /* AD Equalizer Band 7 Coefficient A2 */ + { 0xAE, 0x00 }, /* AE Equalizer Band 7 Coefficient A2 */ + { 0xAF, 0x00 }, /* AF ADC Biquad Coefficient B0 */ + + { 0xB0, 0x00 }, /* B0 ADC Biquad Coefficient B0 */ + { 0xB1, 0x00 }, /* B1 ADC Biquad Coefficient B0 */ + { 0xB2, 0x00 }, /* B2 ADC Biquad Coefficient B1 */ + { 0xB3, 0x00 }, /* B3 ADC Biquad Coefficient B1 */ + { 0xB4, 0x00 }, /* B4 ADC Biquad Coefficient B1 */ + { 0xB5, 0x00 }, /* B5 ADC Biquad Coefficient B2 */ + { 0xB6, 0x00 }, /* B6 ADC Biquad Coefficient B2 */ + { 0xB7, 0x00 }, /* B7 ADC Biquad Coefficient B2 */ + { 0xB8, 0x00 }, /* B8 ADC Biquad Coefficient A1 */ + { 0xB9, 0x00 }, /* B9 ADC Biquad Coefficient A1 */ + { 0xBA, 0x00 }, /* BA ADC Biquad Coefficient A1 */ + { 0xBB, 0x00 }, /* BB ADC Biquad Coefficient A2 */ + { 0xBC, 0x00 }, /* BC ADC Biquad Coefficient A2 */ + { 0xBD, 0x00 }, /* BD ADC Biquad Coefficient A2 */ + { 0xBE, 0x00 }, /* BE Digital Mic 3 Volume */ + { 0xBF, 0x00 }, /* BF Digital Mic 4 Volume */ + + { 0xC0, 0x00 }, /* C0 Digital Mic 34 Biquad Pre Atten */ + { 0xC1, 0x00 }, /* C1 Record TDM Slot */ + { 0xC2, 0x00 }, /* C2 Sample Rate */ + { 0xC3, 0x00 }, /* C3 Digital Mic 34 Biquad Coefficient C3 */ + { 0xC4, 0x00 }, /* C4 Digital Mic 34 Biquad Coefficient C4 */ + { 0xC5, 0x00 }, /* C5 Digital Mic 34 Biquad Coefficient C5 */ + { 0xC6, 0x00 }, /* C6 Digital Mic 34 Biquad Coefficient C6 */ + { 0xC7, 0x00 }, /* C7 Digital Mic 34 Biquad Coefficient C7 */ + { 0xC8, 0x00 }, /* C8 Digital Mic 34 Biquad Coefficient C8 */ + { 0xC9, 0x00 }, /* C9 Digital Mic 34 Biquad Coefficient C9 */ + { 0xCA, 0x00 }, /* CA Digital Mic 34 Biquad Coefficient CA */ + { 0xCB, 0x00 }, /* CB Digital Mic 34 Biquad Coefficient CB */ + { 0xCC, 0x00 }, /* CC Digital Mic 34 Biquad Coefficient CC */ + { 0xCD, 0x00 }, /* CD Digital Mic 34 Biquad Coefficient CD */ + { 0xCE, 0x00 }, /* CE Digital Mic 34 Biquad Coefficient CE */ + { 0xCF, 0x00 }, /* CF Digital Mic 34 Biquad Coefficient CF */ + + { 0xD0, 0x00 }, /* D0 Digital Mic 34 Biquad Coefficient D0 */ + { 0xD1, 0x00 }, /* D1 Digital Mic 34 Biquad Coefficient D1 */ +}; + +static bool max98090_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_REVISION_ID: + return true; + default: + return false; + } +} + +static bool max98090_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_DEVICE_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_INTERRUPT_S: + case M98090_REG_RESERVED: + case M98090_REG_LINE_INPUT_CONFIG: + case M98090_REG_LINE_INPUT_LEVEL: + case M98090_REG_INPUT_MODE: + case M98090_REG_MIC1_INPUT_LEVEL: + case M98090_REG_MIC2_INPUT_LEVEL: + case M98090_REG_MIC_BIAS_VOLTAGE: + case M98090_REG_DIGITAL_MIC_ENABLE: + case M98090_REG_DIGITAL_MIC_CONFIG: + case M98090_REG_LEFT_ADC_MIXER: + case M98090_REG_RIGHT_ADC_MIXER: + case M98090_REG_LEFT_ADC_LEVEL: + case M98090_REG_RIGHT_ADC_LEVEL: + case M98090_REG_ADC_BIQUAD_LEVEL: + case M98090_REG_ADC_SIDETONE: + case M98090_REG_SYSTEM_CLOCK: + case M98090_REG_CLOCK_MODE: + case M98090_REG_CLOCK_RATIO_NI_MSB: + case M98090_REG_CLOCK_RATIO_NI_LSB: + case M98090_REG_CLOCK_RATIO_MI_MSB: + case M98090_REG_CLOCK_RATIO_MI_LSB: + case M98090_REG_MASTER_MODE: + case M98090_REG_INTERFACE_FORMAT: + case M98090_REG_TDM_CONTROL: + case M98090_REG_TDM_FORMAT: + case M98090_REG_IO_CONFIGURATION: + case M98090_REG_FILTER_CONFIG: + case M98090_REG_DAI_PLAYBACK_LEVEL: + case M98090_REG_DAI_PLAYBACK_LEVEL_EQ: + case M98090_REG_LEFT_HP_MIXER: + case M98090_REG_RIGHT_HP_MIXER: + case M98090_REG_HP_CONTROL: + case M98090_REG_LEFT_HP_VOLUME: + case M98090_REG_RIGHT_HP_VOLUME: + case M98090_REG_LEFT_SPK_MIXER: + case M98090_REG_RIGHT_SPK_MIXER: + case M98090_REG_SPK_CONTROL: + case M98090_REG_LEFT_SPK_VOLUME: + case M98090_REG_RIGHT_SPK_VOLUME: + case M98090_REG_DRC_TIMING: + case M98090_REG_DRC_COMPRESSOR: + case M98090_REG_DRC_EXPANDER: + case M98090_REG_DRC_GAIN: + case M98090_REG_RCV_LOUTL_MIXER: + case M98090_REG_RCV_LOUTL_CONTROL: + case M98090_REG_RCV_LOUTL_VOLUME: + case M98090_REG_LOUTR_MIXER: + case M98090_REG_LOUTR_CONTROL: + case M98090_REG_LOUTR_VOLUME: + case M98090_REG_JACK_DETECT: + case M98090_REG_INPUT_ENABLE: + case M98090_REG_OUTPUT_ENABLE: + case M98090_REG_LEVEL_CONTROL: + case M98090_REG_DSP_FILTER_ENABLE: + case M98090_REG_BIAS_CONTROL: + case M98090_REG_DAC_CONTROL: + case M98090_REG_ADC_CONTROL: + case M98090_REG_DEVICE_SHUTDOWN: + case M98090_REG_EQUALIZER_BASE ... M98090_REG_EQUALIZER_BASE + 0x68: + case M98090_REG_RECORD_BIQUAD_BASE ... M98090_REG_RECORD_BIQUAD_BASE + 0x0E: + case M98090_REG_DMIC3_VOLUME: + case M98090_REG_DMIC4_VOLUME: + case M98090_REG_DMIC34_BQ_PREATTEN: + case M98090_REG_RECORD_TDM_SLOT: + case M98090_REG_SAMPLE_RATE: + case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + return true; + default: + return false; + } +} + +static int max98090_reset(struct max98090_priv *max98090) +{ + int ret; + + /* Reset the codec by writing to this write-only reset register */ + ret = regmap_write(max98090->regmap, M98090_REG_SOFTWARE_RESET, + M98090_SWRESET_MASK); + if (ret < 0) { + dev_err(max98090->codec->dev, + "Failed to reset codec: %d\n", ret); + return ret; + } + + msleep(20); + return ret; +} + +static const unsigned int max98090_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_mic_tlv, 0, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_line_single_ended_tlv, + -600, 600, 0); + +static const unsigned int max98090_line_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-600, 300, 0), + 4, 5, TLV_DB_SCALE_ITEM(1400, 600, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_avg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_av_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_dv_tlv, -1500, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_sidetone_tlv, -6050, 200, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_alc_tlv, -1500, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcmakeup_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alccomp_tlv, -3100, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_drcexp_tlv, -6600, 100, 0); + +static const unsigned int max98090_mixout_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(-1200, 250, 0), + 2, 3, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +static const unsigned int max98090_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98090_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 4, TLV_DB_SCALE_ITEM(-4800, 400, 0), + 5, 10, TLV_DB_SCALE_ITEM(-2900, 300, 0), + 11, 14, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 15, 29, TLV_DB_SCALE_ITEM(-500, 100, 0), + 30, 39, TLV_DB_SCALE_ITEM(950, 50, 0), +}; + +static const unsigned int max98090_rcv_lout_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + if (val >= 1) { + /* If on, return the volume */ + val = val - 1; + *select = val; + } else { + /* If off, return last stored value */ + val = *select; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_MIC1_INPUT_LEVEL: + select = &(max98090->pa1en); + break; + case M98090_REG_MIC2_INPUT_LEVEL: + select = &(max98090->pa2en); + break; + case M98090_REG_ADC_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + *select = sel; + + /* Setting a volume is only valid if it is already On */ + if (val >= 1) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, mc->reg, + mask << mc->shift, + sel << mc->shift); + + return 0; +} + +static const char * max98090_perf_pwr_text[] = + { "High Performance", "Low Power" }; +static const char * max98090_pwr_perf_text[] = + { "Low Power", "High Performance" }; + +static const struct soc_enum max98090_vcmbandgap_enum = + SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; + +static const struct soc_enum max98090_osr128_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, + ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); + +static const char *max98090_mode_text[] = { "Voice", "Music" }; + +static const struct soc_enum max98090_mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const struct soc_enum max98090_filter_dmic34mode_enum = + SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34MODE_SHIFT, + ARRAY_SIZE(max98090_mode_text), max98090_mode_text); + +static const char * max98090_drcatk_text[] = + { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; + +static const struct soc_enum max98090_drcatk_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, + ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); + +static const char * max98090_drcrls_text[] = + { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; + +static const struct soc_enum max98090_drcrls_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, + ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); + +static const char * max98090_alccmp_text[] = + { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; + +static const struct soc_enum max98090_alccmp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, + ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); + +static const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; + +static const struct soc_enum max98090_drcexp_enum = + SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, + ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text); + +static const struct soc_enum max98090_dac_perfmode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT, + ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); + +static const struct soc_enum max98090_dachp_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_adchp_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct snd_kcontrol_new max98090_snd_controls[] = { + SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum), + + SOC_SINGLE("DMIC MIC Comp Filter Config", M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0), + + SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", + M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + M98090_MIC_PA1EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_EXT_TLV("MIC2 Boost Volume", + M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + M98090_MIC_PA2EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_TLV("MIC1 Volume", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PGAM1_SHIFT, M98090_MIC_PGAM1_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_TLV("MIC2 Volume", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PGAM2_SHIFT, M98090_MIC_PGAM2_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG135_SHIFT, 0, + M98090_MIXG135_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Single Ended Volume", + M98090_REG_LINE_INPUT_LEVEL, M98090_MIXG246_SHIFT, 0, + M98090_MIXG246_NUM - 1, 1, max98090_line_single_ended_tlv), + + SOC_SINGLE_RANGE_TLV("LINEA Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINAPGA_SHIFT, 0, M98090_LINAPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Volume", M98090_REG_LINE_INPUT_LEVEL, + M98090_LINBPGA_SHIFT, 0, M98090_LINBPGA_NUM - 1, 1, + max98090_line_tlv), + + SOC_SINGLE("LINEA Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFA_SHIFT, M98090_EXTBUFA_NUM - 1, 0), + SOC_SINGLE("LINEB Ext Resistor Gain Mode", M98090_REG_INPUT_MODE, + M98090_EXTBUFB_SHIFT, M98090_EXTBUFB_NUM - 1, 0), + + SOC_SINGLE_TLV("ADCL Boost Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVLG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("ADCR Boost Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVRG_SHIFT, M98090_AVLG_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("ADCL Volume", M98090_REG_LEFT_ADC_LEVEL, + M98090_AVL_SHIFT, M98090_AVL_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("ADCR Volume", M98090_REG_RIGHT_ADC_LEVEL, + M98090_AVR_SHIFT, M98090_AVR_NUM - 1, 1, + max98090_av_tlv), + + SOC_ENUM("ADC Oversampling Rate", max98090_osr128_enum), + SOC_SINGLE("ADC Quantizer Dither", M98090_REG_ADC_CONTROL, + M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0), + SOC_ENUM("ADC High Performance Mode", max98090_adchp_enum), + + SOC_SINGLE("DAC Mono Mode", M98090_REG_IO_CONFIGURATION, + M98090_DMONO_SHIFT, M98090_DMONO_NUM - 1, 0), + SOC_SINGLE("SDIN Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, M98090_SDIEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Mode", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, M98090_SDOEN_NUM - 1, 0), + SOC_SINGLE("SDOUT Hi-Z Mode", M98090_REG_IO_CONFIGURATION, + M98090_HIZOFF_SHIFT, M98090_HIZOFF_NUM - 1, 1), + SOC_ENUM("Filter Mode", max98090_mode_enum), + SOC_SINGLE("Record Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, M98090_AHPF_NUM - 1, 0), + SOC_SINGLE("Playback Path DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_DHPF_SHIFT, M98090_DHPF_NUM - 1, 0), + SOC_SINGLE_TLV("Digital BQ Volume", M98090_REG_ADC_BIQUAD_LEVEL, + M98090_AVBQ_SHIFT, M98090_AVBQ_NUM - 1, 1, max98090_dv_tlv), + SOC_SINGLE_EXT_TLV("Digital Sidetone Volume", + M98090_REG_ADC_SIDETONE, M98090_DVST_SHIFT, + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + SOC_SINGLE_TLV("Digital Coarse Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVG_SHIFT, M98090_DVG_NUM - 1, 0, + max98090_dvg_tlv), + SOC_SINGLE_TLV("Digital Volume", M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DV_SHIFT, M98090_DV_NUM - 1, 1, + max98090_dv_tlv), + SND_SOC_BYTES("EQ Coefficients", M98090_REG_EQUALIZER_BASE, 105), + SOC_SINGLE("Digital EQ 3 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 5 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 7 Band Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ Clipping Detection", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_EQCLPN_SHIFT, M98090_EQCLPN_NUM - 1, + 1), + SOC_SINGLE_TLV("Digital EQ Volume", M98090_REG_DAI_PLAYBACK_LEVEL_EQ, + M98090_DVEQ_SHIFT, M98090_DVEQ_NUM - 1, 1, + max98090_dv_tlv), + + SOC_SINGLE("ALC Enable", M98090_REG_DRC_TIMING, + M98090_DRCEN_SHIFT, M98090_DRCEN_NUM - 1, 0), + SOC_ENUM("ALC Attack Time", max98090_drcatk_enum), + SOC_ENUM("ALC Release Time", max98090_drcrls_enum), + SOC_SINGLE_TLV("ALC Make Up Volume", M98090_REG_DRC_GAIN, + M98090_DRCG_SHIFT, M98090_DRCG_NUM - 1, 0, + max98090_alcmakeup_tlv), + SOC_ENUM("ALC Compression Ratio", max98090_alccmp_enum), + SOC_ENUM("ALC Expansion Ratio", max98090_drcexp_enum), + SOC_SINGLE_TLV("ALC Compression Threshold Volume", + M98090_REG_DRC_COMPRESSOR, M98090_DRCTHC_SHIFT, + M98090_DRCTHC_NUM - 1, 1, max98090_alccomp_tlv), + SOC_SINGLE_TLV("ALC Expansion Threshold Volume", + M98090_REG_DRC_EXPANDER, M98090_DRCTHE_SHIFT, + M98090_DRCTHE_NUM - 1, 1, max98090_drcexp_tlv), + + SOC_ENUM("DAC HP Playback Performance Mode", + max98090_dac_perfmode_enum), + SOC_ENUM("DAC High Performance Mode", max98090_dachp_enum), + + SOC_SINGLE_TLV("Headphone Left Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPLG_SHIFT, + M98090_MIXHPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Headphone Right Mixer Volume", + M98090_REG_HP_CONTROL, M98090_MIXHPRG_SHIFT, + M98090_MIXHPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Speaker Left Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPLG_SHIFT, + M98090_MIXSPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Speaker Right Mixer Volume", + M98090_REG_SPK_CONTROL, M98090_MIXSPRG_SHIFT, + M98090_MIXSPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Receiver Left Mixer Volume", + M98090_REG_RCV_LOUTL_CONTROL, M98090_MIXRCVLG_SHIFT, + M98090_MIXRCVLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Receiver Right Mixer Volume", + M98090_REG_LOUTR_CONTROL, M98090_MIXRCVRG_SHIFT, + M98090_MIXRCVRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_DOUBLE_R_TLV("Headphone Volume", M98090_REG_LEFT_HP_VOLUME, + M98090_REG_RIGHT_HP_VOLUME, M98090_HPVOLL_SHIFT, + M98090_HPVOLL_NUM - 1, 0, max98090_hp_tlv), + + SOC_DOUBLE_R_RANGE_TLV("Speaker Volume", + M98090_REG_LEFT_SPK_VOLUME, M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPVOLL_SHIFT, 24, M98090_SPVOLL_NUM - 1 + 24, + 0, max98090_spk_tlv), + + SOC_DOUBLE_R_TLV("Receiver Volume", M98090_REG_RCV_LOUTL_VOLUME, + M98090_REG_LOUTR_VOLUME, M98090_RCVLVOL_SHIFT, + M98090_RCVLVOL_NUM - 1, 0, max98090_rcv_lout_tlv), + + SOC_SINGLE("Headphone Left Switch", M98090_REG_LEFT_HP_VOLUME, + M98090_HPLM_SHIFT, 1, 1), + SOC_SINGLE("Headphone Right Switch", M98090_REG_RIGHT_HP_VOLUME, + M98090_HPRM_SHIFT, 1, 1), + + SOC_SINGLE("Speaker Left Switch", M98090_REG_LEFT_SPK_VOLUME, + M98090_SPLM_SHIFT, 1, 1), + SOC_SINGLE("Speaker Right Switch", M98090_REG_RIGHT_SPK_VOLUME, + M98090_SPRM_SHIFT, 1, 1), + + SOC_SINGLE("Receiver Left Switch", M98090_REG_RCV_LOUTL_VOLUME, + M98090_RCVLM_SHIFT, 1, 1), + SOC_SINGLE("Receiver Right Switch", M98090_REG_LOUTR_VOLUME, + M98090_RCVRM_SHIFT, 1, 1), + + SOC_SINGLE("Zero-Crossing Detection", M98090_REG_LEVEL_CONTROL, + M98090_ZDENN_SHIFT, M98090_ZDENN_NUM - 1, 1), + SOC_SINGLE("Enhanced Vol Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VS2ENN_SHIFT, M98090_VS2ENN_NUM - 1, 1), + SOC_SINGLE("Volume Adjustment Smoothing", M98090_REG_LEVEL_CONTROL, + M98090_VSENN_SHIFT, M98090_VSENN_NUM - 1, 1), + + SND_SOC_BYTES("Biquad Coefficients", M98090_REG_RECORD_BIQUAD_BASE, 15), + SOC_SINGLE("Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0), +}; + +static const struct snd_kcontrol_new max98091_snd_controls[] = { + + SOC_SINGLE("DMIC34 Zeropad", M98090_REG_SAMPLE_RATE, + M98090_DMIC34_ZEROPAD_SHIFT, + M98090_DMIC34_ZEROPAD_NUM - 1, 0), + + SOC_ENUM("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum), + SOC_SINGLE("DMIC34 DC Blocking", M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34HPF_SHIFT, + M98090_FLT_DMIC34HPF_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC3 Boost Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3G_SHIFT, M98090_DMIC_AV3G_NUM - 1, 0, + max98090_avg_tlv), + SOC_SINGLE_TLV("DMIC4 Boost Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4G_SHIFT, M98090_DMIC_AV4G_NUM - 1, 0, + max98090_avg_tlv), + + SOC_SINGLE_TLV("DMIC3 Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3_SHIFT, M98090_DMIC_AV3_NUM - 1, 1, + max98090_av_tlv), + SOC_SINGLE_TLV("DMIC4 Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4_SHIFT, M98090_DMIC_AV4_NUM - 1, 1, + max98090_av_tlv), + + SND_SOC_BYTES("DMIC34 Biquad Coefficients", + M98090_REG_DMIC34_BIQUAD_BASE, 15), + SOC_SINGLE("DMIC34 Biquad Switch", M98090_REG_DSP_FILTER_ENABLE, + M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC34 BQ PreAttenuation Volume", + M98090_REG_DMIC34_BQ_PREATTEN, M98090_AV34BQ_SHIFT, + M98090_AV34BQ_NUM - 1, 1, max98090_dv_tlv), +}; + +static int max98090_micinput_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, w->reg); + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = (val & M98090_MIC_PA1EN_MASK) >> M98090_MIC_PA1EN_SHIFT; + else + val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; + + + if (val >= 1) { + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { + max98090->pa1en = val - 1; /* Update for volatile */ + } else { + max98090->pa2en = val - 1; /* Update for volatile */ + } + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to most recently selected volume */ + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + val = max98090->pa1en + 1; + else + val = max98090->pa2en + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = 0; + break; + default: + return -EINVAL; + } + + if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA1EN_MASK, + val << M98090_MIC_PA1EN_SHIFT); + else + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA2EN_MASK, + val << M98090_MIC_PA2EN_SHIFT); + + return 0; +} + +static const char *mic1_mux_text[] = { "IN12", "IN56" }; + +static const struct soc_enum mic1_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT, + ARRAY_SIZE(mic1_mux_text), mic1_mux_text); + +static const struct snd_kcontrol_new max98090_mic1_mux = + SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); + +static const char *mic2_mux_text[] = { "IN34", "IN56" }; + +static const struct soc_enum mic2_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT, + ARRAY_SIZE(mic2_mux_text), mic2_mux_text); + +static const struct snd_kcontrol_new max98090_mic2_mux = + SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); + +static const char * max98090_micpre_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_pa1en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const struct soc_enum max98090_pa2en_enum = + SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +/* LINEA mixer switch */ +static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN1SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN3 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN3SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN5 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN5SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN34DIFF_SHIFT, 1, 0), +}; + +/* LINEB mixer switch */ +static const struct snd_kcontrol_new max98090_lineb_mixer_controls[] = { + SOC_DAPM_SINGLE("IN2 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN2SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN4 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN4SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN6 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN6SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LINE_INPUT_CONFIG, + M98090_IN56DIFF_SHIFT, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98090_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_ADC_MIXER, + M98090_MIXADL_MIC2_SHIFT, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_ADC_MIXER, + M98090_MIXADR_MIC2_SHIFT, 1, 0), +}; + +static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; + +static const struct soc_enum ltenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct soc_enum ltenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct snd_kcontrol_new max98090_ltenl_mux = + SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); + +static const struct snd_kcontrol_new max98090_ltenr_mux = + SOC_DAPM_ENUM("LTENR Mux", ltenr_mux_enum); + +static const char *lben_mux_text[] = { "Normal", "Loopback" }; + +static const struct soc_enum lbenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct soc_enum lbenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct snd_kcontrol_new max98090_lbenl_mux = + SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); + +static const struct snd_kcontrol_new max98090_lbenr_mux = + SOC_DAPM_ENUM("LBENR Mux", lbenr_mux_enum); + +static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" }; + +static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" }; + +static const struct soc_enum stenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT, + ARRAY_SIZE(stenl_mux_text), stenl_mux_text); + +static const struct soc_enum stenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT, + ARRAY_SIZE(stenr_mux_text), stenr_mux_text); + +static const struct snd_kcontrol_new max98090_stenl_mux = + SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum); + +static const struct snd_kcontrol_new max98090_stenr_mux = + SOC_DAPM_ENUM("STENR Mux", stenr_mux_enum); + +/* Left speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_SPK_MIXER, + M98090_MIXSPL_MIC2_SHIFT, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_SPK_MIXER, + M98090_MIXSPR_MIC2_SHIFT, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LEFT_HP_MIXER, + M98090_MIXHPL_MIC2_SHIFT, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RIGHT_HP_MIXER, + M98090_MIXHPR_MIC2_SHIFT, 1, 0), +}; + +/* Left receiver mixer switch */ +static const struct snd_kcontrol_new max98090_left_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_RCV_LOUTL_MIXER, + M98090_MIXRCVL_MIC2_SHIFT, 1, 0), +}; + +/* Right receiver mixer switch */ +static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_LOUTR_MIXER, + M98090_MIXRCVR_MIC2_SHIFT, 1, 0), +}; + +static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; + +static const struct soc_enum linmod_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT, + ARRAY_SIZE(linmod_mux_text), linmod_mux_text); + +static const struct snd_kcontrol_new max98090_linmod_mux = + SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); + +static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; + +/* + * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable + */ +static const struct soc_enum mixhplsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhplsel_mux = + SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); + +static const struct soc_enum mixhprsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhprsel_mux = + SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); + +static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMICL"), + SND_SOC_DAPM_INPUT("DMICR"), + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN12"), + SND_SOC_DAPM_INPUT("IN34"), + SND_SOC_DAPM_INPUT("IN56"), + + SND_SOC_DAPM_SUPPLY("MICBIAS", M98090_REG_INPUT_ENABLE, + M98090_MBEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDIEN", M98090_REG_IO_CONFIGURATION, + M98090_SDIEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, + M98090_SDOEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICL_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMICR_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, + M98090_AHPF_SHIFT, 0, NULL, 0), + +/* + * Note: Sysclk and misc power supplies are taken care of by SHDN + */ + + SND_SOC_DAPM_MUX("MIC1 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic1_mux), + + SND_SOC_DAPM_MUX("MIC2 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic2_mux), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PA2EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("LINEA Mixer", SND_SOC_NOPM, 0, 0, + &max98090_linea_mixer_controls[0], + ARRAY_SIZE(max98090_linea_mixer_controls)), + + SND_SOC_DAPM_MIXER("LINEB Mixer", SND_SOC_NOPM, 0, 0, + &max98090_lineb_mixer_controls[0], + ARRAY_SIZE(max98090_lineb_mixer_controls)), + + SND_SOC_DAPM_PGA("LINEA Input", M98090_REG_INPUT_ENABLE, + M98090_LINEAEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("LINEB Input", M98090_REG_INPUT_ENABLE, + M98090_LINEBEN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_adc_mixer_controls[0], + ARRAY_SIZE(max98090_left_adc_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_adc_mixer_controls[0], + ARRAY_SIZE(max98090_right_adc_mixer_controls)), + + SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0), + SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", "HiFi Capture", 1, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("LBENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenl_mux), + + SND_SOC_DAPM_MUX("LBENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenr_mux), + + SND_SOC_DAPM_MUX("LTENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenl_mux), + + SND_SOC_DAPM_MUX("LTENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenr_mux), + + SND_SOC_DAPM_MUX("STENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenl_mux), + + SND_SOC_DAPM_MUX("STENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_stenr_mux), + + SND_SOC_DAPM_AIF_IN("AIFINL", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", "HiFi Playback", 1, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DACL", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DALEN_SHIFT, 0), + SND_SOC_DAPM_DAC("DACR", NULL, M98090_REG_OUTPUT_ENABLE, + M98090_DAREN_SHIFT, 0), + + SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_hp_mixer_controls[0], + ARRAY_SIZE(max98090_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_hp_mixer_controls[0], + ARRAY_SIZE(max98090_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_left_rcv_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_right_rcv_mixer_controls)), + + SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_LOUTR_MIXER, + M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, + M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + + SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_SPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("RCV Left Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("RCV Right Out", M98090_REG_OUTPUT_ENABLE, + M98090_RCVREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RCVL"), + SND_SOC_DAPM_OUTPUT("RCVR"), +}; + +static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + + SND_SOC_DAPM_SUPPLY("DMIC3_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC3_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_DIGITAL_MIC_ENABLE, + M98090_DIGMIC4_SHIFT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { + + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, + + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, + {"DMICL", NULL, "AHPF"}, + {"DMICR", NULL, "AHPF"}, + + /* MIC1 input mux */ + {"MIC1 Mux", "IN12", "IN12"}, + {"MIC1 Mux", "IN56", "IN56"}, + + /* MIC2 input mux */ + {"MIC2 Mux", "IN34", "IN34"}, + {"MIC2 Mux", "IN56", "IN56"}, + + {"MIC1 Input", NULL, "MIC1 Mux"}, + {"MIC2 Input", NULL, "MIC2 Mux"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "IN12 Switch", "IN12"}, + {"Left ADC Mixer", "IN34 Switch", "IN34"}, + {"Left ADC Mixer", "IN56 Switch", "IN56"}, + {"Left ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Left ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "IN12 Switch", "IN12"}, + {"Right ADC Mixer", "IN34 Switch", "IN34"}, + {"Right ADC Mixer", "IN56 Switch", "IN56"}, + {"Right ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Right ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Line A input mixer */ + {"LINEA Mixer", "IN1 Switch", "IN1"}, + {"LINEA Mixer", "IN3 Switch", "IN3"}, + {"LINEA Mixer", "IN5 Switch", "IN5"}, + {"LINEA Mixer", "IN34 Switch", "IN34"}, + + /* Line B input mixer */ + {"LINEB Mixer", "IN2 Switch", "IN2"}, + {"LINEB Mixer", "IN4 Switch", "IN4"}, + {"LINEB Mixer", "IN6 Switch", "IN6"}, + {"LINEB Mixer", "IN56 Switch", "IN56"}, + + {"LINEA Input", NULL, "LINEA Mixer"}, + {"LINEB Input", NULL, "LINEB Mixer"}, + + /* Inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + {"ADCL", NULL, "SHDN"}, + {"ADCR", NULL, "SHDN"}, + + {"LBENL Mux", "Normal", "ADCL"}, + {"LBENL Mux", "Normal", "DMICL"}, + {"LBENL Mux", "Loopback", "LTENL Mux"}, + {"LBENR Mux", "Normal", "ADCR"}, + {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Loopback", "LTENR Mux"}, + + {"AIFOUTL", NULL, "LBENL Mux"}, + {"AIFOUTR", NULL, "LBENR Mux"}, + {"AIFOUTL", NULL, "SHDN"}, + {"AIFOUTR", NULL, "SHDN"}, + {"AIFOUTL", NULL, "SDOEN"}, + {"AIFOUTR", NULL, "SDOEN"}, + + {"LTENL Mux", "Normal", "AIFINL"}, + {"LTENL Mux", "Loopthrough", "LBENL Mux"}, + {"LTENR Mux", "Normal", "AIFINR"}, + {"LTENR Mux", "Loopthrough", "LBENR Mux"}, + + {"DACL", NULL, "LTENL Mux"}, + {"DACR", NULL, "LTENR Mux"}, + + {"STENL Mux", "Sidetone Left", "ADCL"}, + {"STENL Mux", "Sidetone Left", "DMICL"}, + {"STENR Mux", "Sidetone Right", "ADCR"}, + {"STENR Mux", "Sidetone Right", "DMICR"}, + {"DACL", "NULL", "STENL Mux"}, + {"DACR", "NULL", "STENL Mux"}, + + {"AIFINL", NULL, "SHDN"}, + {"AIFINR", NULL, "SHDN"}, + {"AIFINL", NULL, "SDIEN"}, + {"AIFINR", NULL, "SDIEN"}, + {"DACL", NULL, "SHDN"}, + {"DACR", NULL, "SHDN"}, + + /* Left headphone output mixer */ + {"Left Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Left Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Left Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right headphone output mixer */ + {"Right Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Right Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Right Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left speaker output mixer */ + {"Left Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Left Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Left Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right speaker output mixer */ + {"Right Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Right Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Right Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left Receiver output mixer */ + {"Left Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Left Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Left Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right Receiver output mixer */ + {"Right Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Right Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Right Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + {"MIXHPLSEL Mux", "HP Mixer", "Left Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Left Out", NULL, "DACL"}, + {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + + {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Right Out", NULL, "DACR"}, + {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + + {"SPK Left Out", NULL, "Left Speaker Mixer"}, + {"SPK Right Out", NULL, "Right Speaker Mixer"}, + {"RCV Left Out", NULL, "Left Receiver Mixer"}, + + {"LINMOD Mux", "Left and Right", "Right Receiver Mixer"}, + {"LINMOD Mux", "Left Only", "Left Receiver Mixer"}, + {"RCV Right Out", NULL, "LINMOD Mux"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RCVL", NULL, "RCV Left Out"}, + {"RCVR", NULL, "RCV Right Out"}, + +}; + +static const struct snd_soc_dapm_route max98091_dapm_routes[] = { + + /* DMIC inputs */ + {"DMIC3", NULL, "DMIC3_ENA"}, + {"DMIC4", NULL, "DMIC4_ENA"}, + {"DMIC3", NULL, "AHPF"}, + {"DMIC4", NULL, "AHPF"}, + +}; + +static int max98090_add_widgets(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_add_codec_controls(codec, max98090_snd_controls, + ARRAY_SIZE(max98090_snd_controls)); + + if (max98090->devtype == MAX98091) { + snd_soc_add_codec_controls(codec, max98091_snd_controls, + ARRAY_SIZE(max98091_snd_controls)); + } + + snd_soc_dapm_new_controls(dapm, max98090_dapm_widgets, + ARRAY_SIZE(max98090_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98090_dapm_routes, + ARRAY_SIZE(max98090_dapm_routes)); + + if (max98090->devtype == MAX98091) { + snd_soc_dapm_new_controls(dapm, max98091_dapm_widgets, + ARRAY_SIZE(max98091_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, + ARRAY_SIZE(max98091_dapm_routes)); + + } + + return 0; +} + +static const int pclk_rates[] = { + 12000000, 12000000, 13000000, 13000000, + 16000000, 16000000, 19200000, 19200000 +}; + +static const int lrclk_rates[] = { + 8000, 16000, 8000, 16000, + 8000, 16000, 8000, 16000 +}; + +static const int user_pclk_rates[] = { + 13000000, 13000000 +}; + +static const int user_lrclk_rates[] = { + 44100, 48000 +}; + +static const unsigned long long ni_value[] = { + 3528, 768 +}; + +static const unsigned long long mi_value[] = { + 8125, 1625 +}; + +static void max98090_configure_bclk(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned long long ni; + int i; + + if (!max98090->sysclk) { + dev_err(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!max98090->bclk || !max98090->lrclk) { + dev_err(codec->dev, "No audio clocks configured\n"); + return; + } + + /* Skip configuration when operating as slave */ + if (!(snd_soc_read(codec, M98090_REG_MASTER_MODE) & + M98090_MAS_MASK)) { + return; + } + + /* Check for supported PCLK to LRCLK ratios */ + for (i = 0; i < ARRAY_SIZE(pclk_rates); i++) { + if ((pclk_rates[i] == max98090->sysclk) && + (lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found supported PCLK to LRCLK rates 0x%x\n", + i + 0x8); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, + (i + 0x8) << M98090_FREQ_SHIFT); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + return; + } + } + + /* Check for user calculated MI and NI ratios */ + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { + if ((user_pclk_rates[i] == max98090->sysclk) && + (user_lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found user supported PCLK to LRCLK rates\n"); + dev_dbg(codec->dev, "i %d ni %lld mi %lld\n", + i, ni_value[i], mi_value[i]); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, + 1 << M98090_USE_M1_SHIFT); + + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, + ni_value[i] & 0xFF); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_MSB, + (mi_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_MI_LSB, + mi_value[i] & 0xFF); + + return; + } + } + + /* + * Calculate based on MI = 65536 (not as good as either method above) + */ + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_FREQ_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + + /* + * Configure NI when operating as master + * Note: There is a small, but significant audio quality improvement + * by calculating ni and mi. + */ + ni = 65536ULL * (max98090->lrclk < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)max98090->lrclk; + do_div(ni, (unsigned long long int)max98090->sysclk); + dev_info(codec->dev, "No better method found\n"); + dev_info(codec->dev, "Calculating ni %lld with mi 65536\n", ni); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_MSB, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_RATIO_NI_LSB, ni & 0xFF); +} + +static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + u8 regval; + + max98090->dai_fmt = fmt; + cdata = &max98090->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Set to slave mode PLL - MAS mode off */ + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_MSB, 0x00); + snd_soc_write(codec, + M98090_REG_CLOCK_RATIO_NI_LSB, 0x00); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_USE_M1_MASK, 0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + if (max98090->tdm_slots == 4) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_64; + } else if (max98090->tdm_slots == 3) { + /* TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_48; + } else { + /* Few TDM slots, or No TDM */ + regval |= M98090_MAS_MASK | + M98090_BSEL_32; + } + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + snd_soc_write(codec, M98090_REG_MASTER_MODE, regval); + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98090_DLY_MASK; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regval |= M98090_RJ_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Not supported mode */ + default: + dev_err(codec->dev, "DAI format unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98090_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98090_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98090_BCI_MASK|M98090_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + /* + * This accommodates an inverted logic in the MAX98090 chip + * for Bit Clock Invert (BCI). The inverted logic is only + * seen for the case of TDM mode. The remaining cases have + * normal logic. + */ + if (max98090->tdm_slots > 1) { + regval ^= M98090_BCI_MASK; + } + + snd_soc_write(codec, + M98090_REG_INTERFACE_FORMAT, regval); + } + + return 0; +} + +static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + cdata = &max98090->dai[0]; + + if (slots < 0 || slots > 4) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_width = slot_width; + + if (max98090->tdm_slots > 1) { + /* SLOTL SLOTR SLOTDLY */ + snd_soc_write(codec, M98090_REG_TDM_FORMAT, + 0 << M98090_TDM_SLOTL_SHIFT | + 1 << M98090_TDM_SLOTR_SHIFT | + 0 << M98090_TDM_SLOTDLY_SHIFT); + + /* FSW TDM */ + snd_soc_update_bits(codec, M98090_REG_TDM_CONTROL, + M98090_TDM_MASK, + M98090_TDM_MASK); + } + + /* + * Normally advisable to set TDM first, but this permits either order + */ + cdata->fmt = 0; + max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + + return 0; +} + +static int max98090_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { + /* + * Set to normal bias level. + */ + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, + M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); + } + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + /* Set internal pull-up to lowest power mode */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + regcache_mark_dirty(max98090->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const int comp_pclk_rates[] = { + 11289600, 12288000, 12000000, 13000000, 19200000 +}; + +static const int dmic_micclk[] = { + 2, 2, 2, 2, 4, 2 +}; + +static const int comp_lrclk_rates[] = { + 8000, 16000, 32000, 44100, 48000, 96000 +}; + +static const int dmic_comp[6][6] = { + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 1, 1, 1}, + {7, 8, 3, 1, 2, 2}, + {7, 8, 3, 3, 3, 3} +}; + +static int max98090_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int i, j; + + cdata = &max98090->dai[0]; + max98090->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + max98090->bclk *= 2; + + max98090->lrclk = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98090_REG_INTERFACE_FORMAT, + M98090_WS_MASK, 0); + break; + default: + return -EINVAL; + } + + max98090_configure_bclk(codec); + + cdata->rate = max98090->lrclk; + + /* Update filter mode */ + if (max98090->lrclk < 24000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_MODE_MASK, M98090_MODE_MASK); + + /* Update sample rate mode */ + if (max98090->lrclk < 50000) + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG, + M98090_DHF_MASK, M98090_DHF_MASK); + + /* Check for supported PCLK to LRCLK ratios */ + for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) { + if (comp_pclk_rates[j] == max98090->sysclk) { + break; + } + } + + for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { + if (max98090->lrclk <= (comp_lrclk_rates[i] + + comp_lrclk_rates[i + 1]) / 2) { + break; + } + } + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_ENABLE, + M98090_MICCLK_MASK, + dmic_micclk[j] << M98090_MICCLK_SHIFT); + + snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_CONFIG, + M98090_DMIC_COMP_MASK, + dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT); + + return 0; +} + +/* + * PLL / Sysclk + */ +static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + /* Requested clock frequency is already setup */ + if (freq == max98090->sysclk) + return 0; + + /* Setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 40MHz).. + * 0x03 (when master clk is 40MHz to 60MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV1); + } else if ((freq >= 20000000) && (freq < 40000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV2); + } else if ((freq >= 40000000) && (freq < 60000000)) { + snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, + M98090_PSCLK_DIV4); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + max98090->sysclk = freq; + + max98090_configure_bclk(codec); + + return 0; +} + +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int regval; + + regval = mute ? M98090_DVM_MASK : 0; + snd_soc_update_bits(codec, M98090_REG_DAI_PLAYBACK_LEVEL, + M98090_DVM_MASK, regval); + + return 0; +} + +static void max98090_jack_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = container_of(work, + struct max98090_priv, + jack_work.work); + struct snd_soc_codec *codec = max98090->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int status = 0; + int reg; + + /* Read a second time */ + if (max98090->jack_state == M98090_JACK_STATE_NO_HEADSET) { + + /* Strong pull up allows mic detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, 0); + + msleep(50); + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + /* Weak pull up allows only insertion detection */ + snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, + M98090_JDWK_MASK, M98090_JDWK_MASK); + } else { + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + } + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { + case M98090_LSNS_MASK | M98090_JKSNS_MASK: + dev_dbg(codec->dev, "No Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + status |= 0; + + break; + + case 0: + if (max98090->jack_state == + M98090_JACK_STATE_HEADSET) { + + dev_dbg(codec->dev, + "Headset Button Down Detected\n"); + + /* + * max98090_headset_button_event(codec) + * could be defined, then called here. + */ + + status |= SND_JACK_HEADSET; + status |= SND_JACK_BTN_0; + + break; + } + + /* Line is reported as Headphone */ + /* Nokia Headset is reported as Headphone */ + /* Mono Headphone is reported as Headphone */ + dev_dbg(codec->dev, "Headphone Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADPHONE; + + status |= SND_JACK_HEADPHONE; + + break; + + case M98090_JKSNS_MASK: + dev_dbg(codec->dev, "Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADSET; + + status |= SND_JACK_HEADSET; + + break; + + default: + dev_dbg(codec->dev, "Unrecognized Jack Status\n"); + break; + } + + snd_soc_jack_report(max98090->jack, status, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + snd_soc_dapm_sync(dapm); +} + +static irqreturn_t max98090_interrupt(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int mask; + unsigned int active; + + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); + + ret = regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_INTERRUPT_S: %d\n", + ret); + return IRQ_NONE; + } + + ret = regmap_read(max98090->regmap, M98090_REG_DEVICE_STATUS, &active); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_DEVICE_STATUS: %d\n", + ret); + return IRQ_NONE; + } + + dev_dbg(codec->dev, "active=0x%02x mask=0x%02x -> active=0x%02x\n", + active, mask, active & mask); + + active &= mask; + + if (!active) + return IRQ_NONE; + + if (active & M98090_CLD_MASK) { + dev_err(codec->dev, "M98090_CLD_MASK\n"); + } + + if (active & M98090_SLD_MASK) { + dev_dbg(codec->dev, "M98090_SLD_MASK\n"); + } + + if (active & M98090_ULK_MASK) { + dev_err(codec->dev, "M98090_ULK_MASK\n"); + } + + if (active & M98090_JDET_MASK) { + dev_dbg(codec->dev, "M98090_JDET_MASK\n"); + + pm_wakeup_event(codec->dev, 100); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + } + + if (active & M98090_DRCACT_MASK) { + dev_dbg(codec->dev, "M98090_DRCACT_MASK\n"); + } + + if (active & M98090_DRCCLP_MASK) { + dev_err(codec->dev, "M98090_DRCCLP_MASK\n"); + } + + return IRQ_HANDLED; +} + +/** + * max98090_mic_detect - Enable microphone detection via the MAX98090 IRQ + * + * @codec: MAX98090 codec + * @jack: jack to report detection events on + * + * Enable microphone detection via IRQ on the MAX98090. If GPIOs are + * being used to bring out signals to the processor then only platform + * data configuration is needed for MAX98090 and processor GPIOs should + * be configured using snd_soc_jack_add_gpios() instead. + * + * If no jack is supplied detection will be disabled. + */ +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "max98090_mic_detect\n"); + + max98090->jack = jack; + if (jack) { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 1 << M98090_IJDET_SHIFT); + } else { + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IJDET_MASK, + 0); + } + + /* Send an initial empty report */ + snd_soc_jack_report(max98090->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + + return 0; +} +EXPORT_SYMBOL_GPL(max98090_mic_detect); + +#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98090_dai_ops = { + .set_sysclk = max98090_dai_set_sysclk, + .set_fmt = max98090_dai_set_fmt, + .set_tdm_slot = max98090_set_tdm_slot, + .hw_params = max98090_dai_hw_params, + .digital_mute = max98090_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98090_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .ops = &max98090_dai_ops, +} +}; + +static void max98090_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_pdata *pdata = max98090->pdata; + + if (!pdata) { + dev_err(codec->dev, "No platform data\n"); + return; + } + +} + +static int max98090_probe(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int ret = 0; + + dev_dbg(codec->dev, "max98090_probe\n"); + + max98090->codec = codec; + + codec->control_data = max98090->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Reset the codec, the DSP core, and disable all interrupts */ + max98090_reset(max98090); + + /* Initialize private data */ + + max98090->sysclk = (unsigned)-1; + + cdata = &max98090->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + max98090->lin_state = 0; + max98090->pa1en = 0; + max98090->pa2en = 0; + max98090->extmic_mux = 0; + + ret = snd_soc_read(codec, M98090_REG_REVISION_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + + if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { + max98090->devtype = MAX98090; + dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); + } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { + max98090->devtype = MAX98091; + dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); + } else { + max98090->devtype = MAX98090; + dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); + } + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + + /* Enable jack detection */ + snd_soc_write(codec, M98090_REG_JACK_DETECT, + M98090_JDETEN_MASK | M98090_JDEB_25MS); + + /* Register for interrupts */ + dev_dbg(codec->dev, "irq = %d\n", max98090->irq); + + ret = request_threaded_irq(max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING, + "max98090_interrupt", codec); + if (ret < 0) { + dev_err(codec->dev, "request_irq failed: %d\n", + ret); + } + + /* + * Clear any old interrupts. + * An old interrupt ocurring prior to installing the ISR + * can keep a new interrupt from generating a trigger. + */ + snd_soc_read(codec, M98090_REG_DEVICE_STATUS); + + /* High Performance is default */ + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_DACHP_MASK, + 1 << M98090_DACHP_SHIFT); + snd_soc_update_bits(codec, M98090_REG_DAC_CONTROL, + M98090_PERFMODE_MASK, + 0 << M98090_PERFMODE_SHIFT); + snd_soc_update_bits(codec, M98090_REG_ADC_CONTROL, + M98090_ADCHP_MASK, + 1 << M98090_ADCHP_SHIFT); + + /* Turn on VCM bandgap reference */ + snd_soc_write(codec, M98090_REG_BIAS_CONTROL, + M98090_VCM_MODE_MASK); + + max98090_handle_pdata(codec); + + max98090_add_widgets(codec); + +err_access: + return ret; +} + +static int max98090_remove(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&max98090->jack_work); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98090 = { + .probe = max98090_probe, + .remove = max98090_remove, + .set_bias_level = max98090_set_bias_level, +}; + +static const struct regmap_config max98090_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX98090_MAX_REGISTER, + .reg_defaults = max98090_reg, + .num_reg_defaults = ARRAY_SIZE(max98090_reg), + .volatile_reg = max98090_volatile_register, + .readable_reg = max98090_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98090_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98090_priv *max98090; + int ret; + + pr_debug("max98090_i2c_probe\n"); + + max98090 = devm_kzalloc(&i2c->dev, sizeof(struct max98090_priv), + GFP_KERNEL); + if (max98090 == NULL) + return -ENOMEM; + + max98090->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98090); + max98090->control_data = i2c; + max98090->pdata = i2c->dev.platform_data; + max98090->irq = i2c->irq; + + max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + if (IS_ERR(max98090->regmap)) { + ret = PTR_ERR(max98090->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98090, max98090_dai, + ARRAY_SIZE(max98090_dai)); + if (ret < 0) + regmap_exit(max98090->regmap); + +err_enable: + return ret; +} + +static int max98090_i2c_remove(struct i2c_client *client) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); + snd_soc_unregister_codec(&client->dev); + regmap_exit(max98090->regmap); + return 0; +} + +static int max98090_runtime_resume(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, false); + + regcache_sync(max98090->regmap); + + return 0; +} + +static int max98090_runtime_suspend(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, true); + + return 0; +} + +static struct dev_pm_ops max98090_pm = { + SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + max98090_runtime_resume, NULL) +}; + +static const struct i2c_device_id max98090_i2c_id[] = { + { "max98090", MAX98090 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); + +static struct i2c_driver max98090_i2c_driver = { + .driver = { + .name = "max98090", + .owner = THIS_MODULE, + .pm = &max98090_pm, + }, + .probe = max98090_i2c_probe, + .remove = max98090_i2c_remove, + .id_table = max98090_i2c_id, +}; + +module_i2c_driver(max98090_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); +MODULE_AUTHOR("Peter Hsiang, Jesse Marroqin, Jerry Wong"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h new file mode 100755 index 0000000..6c9fa34 --- /dev/null +++ b/sound/soc/codecs/max98090.h @@ -0,0 +1,1549 @@ +/* + * max98090.h -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98090_H +#define _MAX98090_H + +#include <linux/version.h> + +/* One can override the Linux version here with an explicit version number */ +#define M98090_LINUX_VERSION LINUX_VERSION_CODE + +/* + * MAX98090 Register Definitions + */ + +#define M98090_REG_SOFTWARE_RESET 0x00 +#define M98090_REG_DEVICE_STATUS 0x01 +#define M98090_REG_JACK_STATUS 0x02 +#define M98090_REG_INTERRUPT_S 0x03 +#define M98090_REG_QUICK_SYSTEM_CLOCK 0x04 +#define M98090_REG_QUICK_SAMPLE_RATE 0x05 +#define M98090_REG_DAI_INTERFACE 0x06 +#define M98090_REG_DAC_PATH 0x07 +#define M98090_REG_MIC_DIRECT_TO_ADC 0x08 +#define M98090_REG_LINE_TO_ADC 0x09 +#define M98090_REG_ANALOG_MIC_LOOP 0x0A +#define M98090_REG_ANALOG_LINE_LOOP 0x0B +#define M98090_REG_RESERVED 0x0C +#define M98090_REG_LINE_INPUT_CONFIG 0x0D +#define M98090_REG_LINE_INPUT_LEVEL 0x0E +#define M98090_REG_INPUT_MODE 0x0F +#define M98090_REG_MIC1_INPUT_LEVEL 0x10 +#define M98090_REG_MIC2_INPUT_LEVEL 0x11 +#define M98090_REG_MIC_BIAS_VOLTAGE 0x12 +#define M98090_REG_DIGITAL_MIC_ENABLE 0x13 +#define M98090_REG_DIGITAL_MIC_CONFIG 0x14 +#define M98090_REG_LEFT_ADC_MIXER 0x15 +#define M98090_REG_RIGHT_ADC_MIXER 0x16 +#define M98090_REG_LEFT_ADC_LEVEL 0x17 +#define M98090_REG_RIGHT_ADC_LEVEL 0x18 +#define M98090_REG_ADC_BIQUAD_LEVEL 0x19 +#define M98090_REG_ADC_SIDETONE 0x1A +#define M98090_REG_SYSTEM_CLOCK 0x1B +#define M98090_REG_CLOCK_MODE 0x1C +#define M98090_REG_CLOCK_RATIO_NI_MSB 0x1D +#define M98090_REG_CLOCK_RATIO_NI_LSB 0x1E +#define M98090_REG_CLOCK_RATIO_MI_MSB 0x1F +#define M98090_REG_CLOCK_RATIO_MI_LSB 0x20 +#define M98090_REG_MASTER_MODE 0x21 +#define M98090_REG_INTERFACE_FORMAT 0x22 +#define M98090_REG_TDM_CONTROL 0x23 +#define M98090_REG_TDM_FORMAT 0x24 +#define M98090_REG_IO_CONFIGURATION 0x25 +#define M98090_REG_FILTER_CONFIG 0x26 +#define M98090_REG_DAI_PLAYBACK_LEVEL 0x27 +#define M98090_REG_DAI_PLAYBACK_LEVEL_EQ 0x28 +#define M98090_REG_LEFT_HP_MIXER 0x29 +#define M98090_REG_RIGHT_HP_MIXER 0x2A +#define M98090_REG_HP_CONTROL 0x2B +#define M98090_REG_LEFT_HP_VOLUME 0x2C +#define M98090_REG_RIGHT_HP_VOLUME 0x2D +#define M98090_REG_LEFT_SPK_MIXER 0x2E +#define M98090_REG_RIGHT_SPK_MIXER 0x2F +#define M98090_REG_SPK_CONTROL 0x30 +#define M98090_REG_LEFT_SPK_VOLUME 0x31 +#define M98090_REG_RIGHT_SPK_VOLUME 0x32 +#define M98090_REG_DRC_TIMING 0x33 +#define M98090_REG_DRC_COMPRESSOR 0x34 +#define M98090_REG_DRC_EXPANDER 0x35 +#define M98090_REG_DRC_GAIN 0x36 +#define M98090_REG_RCV_LOUTL_MIXER 0x37 +#define M98090_REG_RCV_LOUTL_CONTROL 0x38 +#define M98090_REG_RCV_LOUTL_VOLUME 0x39 +#define M98090_REG_LOUTR_MIXER 0x3A +#define M98090_REG_LOUTR_CONTROL 0x3B +#define M98090_REG_LOUTR_VOLUME 0x3C +#define M98090_REG_JACK_DETECT 0x3D +#define M98090_REG_INPUT_ENABLE 0x3E +#define M98090_REG_OUTPUT_ENABLE 0x3F +#define M98090_REG_LEVEL_CONTROL 0x40 +#define M98090_REG_DSP_FILTER_ENABLE 0x41 +#define M98090_REG_BIAS_CONTROL 0x42 +#define M98090_REG_DAC_CONTROL 0x43 +#define M98090_REG_ADC_CONTROL 0x44 +#define M98090_REG_DEVICE_SHUTDOWN 0x45 +#define M98090_REG_EQUALIZER_BASE 0x46 +#define M98090_REG_RECORD_BIQUAD_BASE 0xAF +#define M98090_REG_DMIC3_VOLUME 0xBE +#define M98090_REG_DMIC4_VOLUME 0xBF +#define M98090_REG_DMIC34_BQ_PREATTEN 0xC0 +#define M98090_REG_RECORD_TDM_SLOT 0xC1 +#define M98090_REG_SAMPLE_RATE 0xC2 +#define M98090_REG_DMIC34_BIQUAD_BASE 0xC3 +#define M98090_REG_REVISION_ID 0xFF + +#define M98090_REG_CNT (0xFF+1) +#define MAX98090_MAX_REGISTER 0xFF + +/* MAX98090 Register Bit Fields */ + +/* + * M98090_REG_SOFTWARE_RESET + */ +#define M98090_SWRESET_MASK (1<<7) +#define M98090_SWRESET_SHIFT 7 +#define M98090_SWRESET_WIDTH 1 + +/* + * M98090_REG_DEVICE_STATUS + */ +#define M98090_CLD_MASK (1<<7) +#define M98090_CLD_SHIFT 7 +#define M98090_CLD_WIDTH 1 +#define M98090_SLD_MASK (1<<6) +#define M98090_SLD_SHIFT 6 +#define M98090_SLD_WIDTH 1 +#define M98090_ULK_MASK (1<<5) +#define M98090_ULK_SHIFT 5 +#define M98090_ULK_WIDTH 1 +#define M98090_JDET_MASK (1<<2) +#define M98090_JDET_SHIFT 2 +#define M98090_JDET_WIDTH 1 +#define M98090_DRCACT_MASK (1<<1) +#define M98090_DRCACT_SHIFT 1 +#define M98090_DRCACT_WIDTH 1 +#define M98090_DRCCLP_MASK (1<<0) +#define M98090_DRCCLP_SHIFT 0 +#define M98090_DRCCLP_WIDTH 1 + +/* + * M98090_REG_JACK_STATUS + */ +#define M98090_LSNS_MASK (1<<2) +#define M98090_LSNS_SHIFT 2 +#define M98090_LSNS_WIDTH 1 +#define M98090_JKSNS_MASK (1<<1) +#define M98090_JKSNS_SHIFT 1 +#define M98090_JKSNS_WIDTH 1 + +/* + * M98090_REG_INTERRUPT_S + */ +#define M98090_ICLD_MASK (1<<7) +#define M98090_ICLD_SHIFT 7 +#define M98090_ICLD_WIDTH 1 +#define M98090_ISLD_MASK (1<<6) +#define M98090_ISLD_SHIFT 6 +#define M98090_ISLD_WIDTH 1 +#define M98090_IULK_MASK (1<<5) +#define M98090_IULK_SHIFT 5 +#define M98090_IULK_WIDTH 1 +#define M98090_IJDET_MASK (1<<2) +#define M98090_IJDET_SHIFT 2 +#define M98090_IJDET_WIDTH 1 +#define M98090_IDRCACT_MASK (1<<1) +#define M98090_IDRCACT_SHIFT 1 +#define M98090_IDRCACT_WIDTH 1 +#define M98090_IDRCCLP_MASK (1<<0) +#define M98090_IDRCCLP_SHIFT 0 +#define M98090_IDRCCLP_WIDTH 1 + +/* + * M98090_REG_QUICK_SYSTEM_CLOCK + */ +#define M98090_26M_MASK (1<<7) +#define M98090_26M_SHIFT 7 +#define M98090_26M_WIDTH 1 +#define M98090_19P2M_MASK (1<<6) +#define M98090_19P2M_SHIFT 6 +#define M98090_19P2M_WIDTH 1 +#define M98090_13M_MASK (1<<5) +#define M98090_13M_SHIFT 5 +#define M98090_13M_WIDTH 1 +#define M98090_12P288M_MASK (1<<4) +#define M98090_12P288M_SHIFT 4 +#define M98090_12P288M_WIDTH 1 +#define M98090_12M_MASK (1<<3) +#define M98090_12M_SHIFT 3 +#define M98090_12M_WIDTH 1 +#define M98090_11P2896M_MASK (1<<2) +#define M98090_11P2896M_SHIFT 2 +#define M98090_11P2896M_WIDTH 1 +#define M98090_256FS_MASK (1<<0) +#define M98090_256FS_SHIFT 0 +#define M98090_256FS_WIDTH 1 +#define M98090_CLK_ALL_SHIFT 0 +#define M98090_CLK_ALL_WIDTH 8 +#define M98090_CLK_ALL_NUM (1<<M98090_CLK_ALL_WIDTH) + +/* + * M98090_REG_QUICK_SAMPLE_RATE + */ +#define M98090_SR_96K_MASK (1<<5) +#define M98090_SR_96K_SHIFT 5 +#define M98090_SR_96K_WIDTH 1 +#define M98090_SR_32K_MASK (1<<4) +#define M98090_SR_32K_SHIFT 4 +#define M98090_SR_32K_WIDTH 1 +#define M98090_SR_48K_MASK (1<<3) +#define M98090_SR_48K_SHIFT 3 +#define M98090_SR_48K_WIDTH 1 +#define M98090_SR_44K1_MASK (1<<2) +#define M98090_SR_44K1_SHIFT 2 +#define M98090_SR_44K1_WIDTH 1 +#define M98090_SR_16K_MASK (1<<1) +#define M98090_SR_16K_SHIFT 1 +#define M98090_SR_16K_WIDTH 1 +#define M98090_SR_8K_MASK (1<<0) +#define M98090_SR_8K_SHIFT 0 +#define M98090_SR_8K_WIDTH 1 +#define M98090_SR_MASK 0x3F +#define M98090_SR_ALL_SHIFT 0 +#define M98090_SR_ALL_WIDTH 8 +#define M98090_SR_ALL_NUM (1<<M98090_SR_ALL_WIDTH) + +/* + * M98090_REG_DAI_INTERFACE + */ +#define M98090_RJ_M_MASK (1<<5) +#define M98090_RJ_M_SHIFT 5 +#define M98090_RJ_M_WIDTH 1 +#define M98090_RJ_S_MASK (1<<4) +#define M98090_RJ_S_SHIFT 4 +#define M98090_RJ_S_WIDTH 1 +#define M98090_LJ_M_MASK (1<<3) +#define M98090_LJ_M_SHIFT 3 +#define M98090_LJ_M_WIDTH 1 +#define M98090_LJ_S_MASK (1<<2) +#define M98090_LJ_S_SHIFT 2 +#define M98090_LJ_S_WIDTH 1 +#define M98090_I2S_M_MASK (1<<1) +#define M98090_I2S_M_SHIFT 1 +#define M98090_I2S_M_WIDTH 1 +#define M98090_I2S_S_MASK (1<<0) +#define M98090_I2S_S_SHIFT 0 +#define M98090_I2S_S_WIDTH 1 +#define M98090_DAI_ALL_SHIFT 0 +#define M98090_DAI_ALL_WIDTH 8 +#define M98090_DAI_ALL_NUM (1<<M98090_DAI_ALL_WIDTH) + +/* + * M98090_REG_DAC_PATH + */ +#define M98090_DIG2_HP_MASK (1<<7) +#define M98090_DIG2_HP_SHIFT 7 +#define M98090_DIG2_HP_WIDTH 1 +#define M98090_DIG2_EAR_MASK (1<<6) +#define M98090_DIG2_EAR_SHIFT 6 +#define M98090_DIG2_EAR_WIDTH 1 +#define M98090_DIG2_SPK_MASK (1<<5) +#define M98090_DIG2_SPK_SHIFT 5 +#define M98090_DIG2_SPK_WIDTH 1 +#define M98090_DIG2_LOUT_MASK (1<<4) +#define M98090_DIG2_LOUT_SHIFT 4 +#define M98090_DIG2_LOUT_WIDTH 1 +#define M98090_DIG2_ALL_SHIFT 0 +#define M98090_DIG2_ALL_WIDTH 8 +#define M98090_DIG2_ALL_NUM (1<<M98090_DIG2_ALL_WIDTH) + +/* + * M98090_REG_MIC_DIRECT_TO_ADC + */ +#define M98090_IN12_MIC1_MASK (1<<7) +#define M98090_IN12_MIC1_SHIFT 7 +#define M98090_IN12_MIC1_WIDTH 1 +#define M98090_IN34_MIC2_MASK (1<<6) +#define M98090_IN34_MIC2_SHIFT 6 +#define M98090_IN34_MIC2_WIDTH 1 +#define M98090_IN56_MIC1_MASK (1<<5) +#define M98090_IN56_MIC1_SHIFT 5 +#define M98090_IN56_MIC1_WIDTH 1 +#define M98090_IN56_MIC2_MASK (1<<4) +#define M98090_IN56_MIC2_SHIFT 4 +#define M98090_IN56_MIC2_WIDTH 1 +#define M98090_IN12_DADC_MASK (1<<3) +#define M98090_IN12_DADC_SHIFT 3 +#define M98090_IN12_DADC_WIDTH 1 +#define M98090_IN34_DADC_MASK (1<<2) +#define M98090_IN34_DADC_SHIFT 2 +#define M98090_IN34_DADC_WIDTH 1 +#define M98090_IN56_DADC_MASK (1<<1) +#define M98090_IN56_DADC_SHIFT 1 +#define M98090_IN56_DADC_WIDTH 1 +#define M98090_MIC_ALL_SHIFT 0 +#define M98090_MIC_ALL_WIDTH 8 +#define M98090_MIC_ALL_NUM (1<<M98090_MIC_ALL_WIDTH) + +/* + * M98090_REG_LINE_TO_ADC + */ +#define M98090_IN12S_AB_MASK (1<<7) +#define M98090_IN12S_AB_SHIFT 7 +#define M98090_IN12S_AB_WIDTH 1 +#define M98090_IN34S_AB_MASK (1<<6) +#define M98090_IN34S_AB_SHIFT 6 +#define M98090_IN34S_AB_WIDTH 1 +#define M98090_IN56S_AB_MASK (1<<5) +#define M98090_IN56S_AB_SHIFT 5 +#define M98090_IN56S_AB_WIDTH 1 +#define M98090_IN34D_A_MASK (1<<4) +#define M98090_IN34D_A_SHIFT 4 +#define M98090_IN34D_A_WIDTH 1 +#define M98090_IN56D_B_MASK (1<<3) +#define M98090_IN56D_B_SHIFT 3 +#define M98090_IN56D_B_WIDTH 1 +#define M98090_LINE_ALL_SHIFT 0 +#define M98090_LINE_ALL_WIDTH 8 +#define M98090_LINE_ALL_NUM (1<<M98090_LINE_ALL_WIDTH) + +/* + * M98090_REG_ANALOG_MIC_LOOP + */ +#define M98090_IN12_M1HPL_MASK (1<<7) +#define M98090_IN12_M1HPL_SHIFT 7 +#define M98090_IN12_M1HPL_WIDTH 1 +#define M98090_IN12_M1SPKL_MASK (1<<6) +#define M98090_IN12_M1SPKL_SHIFT 6 +#define M98090_IN12_M1SPKL_WIDTH 1 +#define M98090_IN12_M1EAR_MASK (1<<5) +#define M98090_IN12_M1EAR_SHIFT 5 +#define M98090_IN12_M1EAR_WIDTH 1 +#define M98090_IN12_M1LOUTL_MASK (1<<4) +#define M98090_IN12_M1LOUTL_SHIFT 4 +#define M98090_IN12_M1LOUTL_WIDTH 1 +#define M98090_IN34_M2HPR_MASK (1<<3) +#define M98090_IN34_M2HPR_SHIFT 3 +#define M98090_IN34_M2HPR_WIDTH 1 +#define M98090_IN34_M2SPKR_MASK (1<<2) +#define M98090_IN34_M2SPKR_SHIFT 2 +#define M98090_IN34_M2SPKR_WIDTH 1 +#define M98090_IN34_M2EAR_MASK (1<<1) +#define M98090_IN34_M2EAR_SHIFT 1 +#define M98090_IN34_M2EAR_WIDTH 1 +#define M98090_IN34_M2LOUTR_MASK (1<<0) +#define M98090_IN34_M2LOUTR_SHIFT 0 +#define M98090_IN34_M2LOUTR_WIDTH 1 +#define M98090_AMIC_ALL_SHIFT 0 +#define M98090_AMIC_ALL_WIDTH 8 +#define M98090_AMIC_ALL_NUM (1<<M98090_AMIC_ALL_WIDTH) + +/* + * M98090_REG_ANALOG_LINE_LOOP + */ +#define M98090_IN12S_ABHP_MASK (1<<7) +#define M98090_IN12S_ABHP_SHIFT 7 +#define M98090_IN12S_ABHP_WIDTH 1 +#define M98090_IN34D_ASPKL_MASK (1<<6) +#define M98090_IN34D_ASPKL_SHIFT 6 +#define M98090_IN34D_ASPKL_WIDTH 1 +#define M98090_IN34D_AEAR_MASK (1<<5) +#define M98090_IN34D_AEAR_SHIFT 5 +#define M98090_IN34D_AEAR_WIDTH 1 +#define M98090_IN12S_ABLOUT_MASK (1<<4) +#define M98090_IN12S_ABLOUT_SHIFT 4 +#define M98090_IN12S_ABLOUT_WIDTH 1 +#define M98090_IN34S_ABHP_MASK (1<<3) +#define M98090_IN34S_ABHP_SHIFT 3 +#define M98090_IN34S_ABHP_WIDTH 1 +#define M98090_IN56D_BSPKR_MASK (1<<2) +#define M98090_IN56D_BSPKR_SHIFT 2 +#define M98090_IN56D_BSPKR_WIDTH 1 +#define M98090_IN56D_BEAR_MASK (1<<1) +#define M98090_IN56D_BEAR_SHIFT 1 +#define M98090_IN56D_BEAR_WIDTH 1 +#define M98090_IN34S_ABLOUT_MASK (1<<0) +#define M98090_IN34S_ABLOUT_SHIFT 0 +#define M98090_IN34S_ABLOUT_WIDTH 1 +#define M98090_ALIN_ALL_SHIFT 0 +#define M98090_ALIN_ALL_WIDTH 8 +#define M98090_ALIN_ALL_NUM (1<<M98090_ALIN_ALL_WIDTH) + +/* + * M98090_REG_RESERVED + */ + +/* + * M98090_REG_LINE_INPUT_CONFIG + */ +#define M98090_IN34DIFF_MASK (1<<7) +#define M98090_IN34DIFF_SHIFT 7 +#define M98090_IN34DIFF_WIDTH 1 +#define M98090_IN56DIFF_MASK (1<<6) +#define M98090_IN56DIFF_SHIFT 6 +#define M98090_IN56DIFF_WIDTH 1 +#define M98090_IN1SEEN_MASK (1<<5) +#define M98090_IN1SEEN_SHIFT 5 +#define M98090_IN1SEEN_WIDTH 1 +#define M98090_IN2SEEN_MASK (1<<4) +#define M98090_IN2SEEN_SHIFT 4 +#define M98090_IN2SEEN_WIDTH 1 +#define M98090_IN3SEEN_MASK (1<<3) +#define M98090_IN3SEEN_SHIFT 3 +#define M98090_IN3SEEN_WIDTH 1 +#define M98090_IN4SEEN_MASK (1<<2) +#define M98090_IN4SEEN_SHIFT 2 +#define M98090_IN4SEEN_WIDTH 1 +#define M98090_IN5SEEN_MASK (1<<1) +#define M98090_IN5SEEN_SHIFT 1 +#define M98090_IN5SEEN_WIDTH 1 +#define M98090_IN6SEEN_MASK (1<<0) +#define M98090_IN6SEEN_SHIFT 0 +#define M98090_IN6SEEN_WIDTH 1 + +/* + * M98090_REG_LINE_INPUT_LEVEL + */ +#define M98090_MIXG135_MASK (1<<7) +#define M98090_MIXG135_SHIFT 7 +#define M98090_MIXG135_WIDTH 1 +#define M98090_MIXG135_NUM (1<<M98090_MIXG135_WIDTH) +#define M98090_MIXG246_MASK (1<<6) +#define M98090_MIXG246_SHIFT 6 +#define M98090_MIXG246_WIDTH 1 +#define M98090_MIXG246_NUM (1<<M98090_MIXG246_WIDTH) +#define M98090_LINAPGA_MASK (7<<3) +#define M98090_LINAPGA_SHIFT 3 +#define M98090_LINAPGA_WIDTH 3 +#define M98090_LINAPGA_NUM 6 +#define M98090_LINBPGA_MASK (7<<0) +#define M98090_LINBPGA_SHIFT 0 +#define M98090_LINBPGA_WIDTH 3 +#define M98090_LINBPGA_NUM 6 + +/* + * M98090_REG_INPUT_MODE + */ +#define M98090_EXTBUFA_MASK (1<<7) +#define M98090_EXTBUFA_SHIFT 7 +#define M98090_EXTBUFA_WIDTH 1 +#define M98090_EXTBUFA_NUM (1<<M98090_EXTBUFA_WIDTH) +#define M98090_EXTBUFB_MASK (1<<6) +#define M98090_EXTBUFB_SHIFT 6 +#define M98090_EXTBUFB_WIDTH 1 +#define M98090_EXTBUFB_NUM (1<<M98090_EXTBUFB_WIDTH) +#define M98090_EXTMIC_MASK (3<<0) +#define M98090_EXTMIC_SHIFT 0 +#define M98090_EXTMIC1_SHIFT 0 +#define M98090_EXTMIC2_SHIFT 1 +#define M98090_EXTMIC_WIDTH 2 +#define M98090_EXTMIC_NONE (0<<0) +#define M98090_EXTMIC_MIC1 (1<<0) +#define M98090_EXTMIC_MIC2 (2<<0) + +/* + * M98090_REG_MIC1_INPUT_LEVEL + */ +#define M98090_MIC_PA1EN_MASK (3<<5) +#define M98090_MIC_PA1EN_SHIFT 5 +#define M98090_MIC_PA1EN_WIDTH 2 +#define M98090_MIC_PA1EN_NUM 3 +#define M98090_MIC_PGAM1_MASK (31<<0) +#define M98090_MIC_PGAM1_SHIFT 0 +#define M98090_MIC_PGAM1_WIDTH 5 +#define M98090_MIC_PGAM1_NUM 21 + +/* + * M98090_REG_MIC2_INPUT_LEVEL + */ +#define M98090_MIC_PA2EN_MASK (3<<5) +#define M98090_MIC_PA2EN_SHIFT 5 +#define M98090_MIC_PA2EN_WIDTH 2 +#define M98090_MIC_PA2EN_NUM 3 +#define M98090_MIC_PGAM2_MASK (31<<0) +#define M98090_MIC_PGAM2_SHIFT 0 +#define M98090_MIC_PGAM2_WIDTH 5 +#define M98090_MIC_PGAM2_NUM 21 + +/* + * M98090_REG_MIC_BIAS_VOLTAGE + */ +#define M98090_MBVSEL_MASK (3<<0) +#define M98090_MBVSEL_SHIFT 0 +#define M98090_MBVSEL_WIDTH 2 +#define M98090_MBVSEL_2V8 (3<<0) +#define M98090_MBVSEL_2V55 (2<<0) +#define M98090_MBVSEL_2V4 (1<<0) +#define M98090_MBVSEL_2V2 (0<<0) + +/* + * M98090_REG_DIGITAL_MIC_ENABLE + */ +#define M98090_MICCLK_MASK (7<<4) +#define M98090_MICCLK_SHIFT 4 +#define M98090_MICCLK_WIDTH 3 +#define M98090_DIGMIC4_MASK (1<<3) +#define M98090_DIGMIC4_SHIFT 3 +#define M98090_DIGMIC4_WIDTH 1 +#define M98090_DIGMIC4_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DIGMIC3_MASK (1<<2) +#define M98090_DIGMIC3_SHIFT 2 +#define M98090_DIGMIC3_WIDTH 1 +#define M98090_DIGMIC3_NUM (1<<M98090_DIGMIC3_WIDTH) +#define M98090_DIGMICR_MASK (1<<1) +#define M98090_DIGMICR_SHIFT 1 +#define M98090_DIGMICR_WIDTH 1 +#define M98090_DIGMICR_NUM (1<<M98090_DIGMICR_WIDTH) +#define M98090_DIGMICL_MASK (1<<0) +#define M98090_DIGMICL_SHIFT 0 +#define M98090_DIGMICL_WIDTH 1 +#define M98090_DIGMICL_NUM (1<<M98090_DIGMICL_WIDTH) + +/* + * M98090_REG_DIGITAL_MIC_CONFIG + */ +#define M98090_DMIC_COMP_MASK (15<<4) +#define M98090_DMIC_COMP_SHIFT 4 +#define M98090_DMIC_COMP_WIDTH 4 +#define M98090_DMIC_COMP_NUM (1<<M98090_DMIC_COMP_WIDTH) +#define M98090_DMIC_FREQ_MASK (3<<0) +#define M98090_DMIC_FREQ_SHIFT 0 +#define M98090_DMIC_FREQ_WIDTH 2 + +/* + * M98090_REG_LEFT_ADC_MIXER + */ +#define M98090_MIXADL_MIC2_MASK (1<<6) +#define M98090_MIXADL_MIC2_SHIFT 6 +#define M98090_MIXADL_MIC2_WIDTH 1 +#define M98090_MIXADL_MIC1_MASK (1<<5) +#define M98090_MIXADL_MIC1_SHIFT 5 +#define M98090_MIXADL_MIC1_WIDTH 1 +#define M98090_MIXADL_LINEB_MASK (1<<4) +#define M98090_MIXADL_LINEB_SHIFT 4 +#define M98090_MIXADL_LINEB_WIDTH 1 +#define M98090_MIXADL_LINEA_MASK (1<<3) +#define M98090_MIXADL_LINEA_SHIFT 3 +#define M98090_MIXADL_LINEA_WIDTH 1 +#define M98090_MIXADL_IN65DIFF_MASK (1<<2) +#define M98090_MIXADL_IN65DIFF_SHIFT 2 +#define M98090_MIXADL_IN65DIFF_WIDTH 1 +#define M98090_MIXADL_IN34DIFF_MASK (1<<1) +#define M98090_MIXADL_IN34DIFF_SHIFT 1 +#define M98090_MIXADL_IN34DIFF_WIDTH 1 +#define M98090_MIXADL_IN12DIFF_MASK (1<<0) +#define M98090_MIXADL_IN12DIFF_SHIFT 0 +#define M98090_MIXADL_IN12DIFF_WIDTH 1 +#define M98090_MIXADL_MASK (255<<0) +#define M98090_MIXADL_SHIFT 0 +#define M98090_MIXADL_WIDTH 8 + +/* + * M98090_REG_RIGHT_ADC_MIXER + */ +#define M98090_MIXADR_MIC2_MASK (1<<6) +#define M98090_MIXADR_MIC2_SHIFT 6 +#define M98090_MIXADR_MIC2_WIDTH 1 +#define M98090_MIXADR_MIC1_MASK (1<<5) +#define M98090_MIXADR_MIC1_SHIFT 5 +#define M98090_MIXADR_MIC1_WIDTH 1 +#define M98090_MIXADR_LINEB_MASK (1<<4) +#define M98090_MIXADR_LINEB_SHIFT 4 +#define M98090_MIXADR_LINEB_WIDTH 1 +#define M98090_MIXADR_LINEA_MASK (1<<3) +#define M98090_MIXADR_LINEA_SHIFT 3 +#define M98090_MIXADR_LINEA_WIDTH 1 +#define M98090_MIXADR_IN65DIFF_MASK (1<<2) +#define M98090_MIXADR_IN65DIFF_SHIFT 2 +#define M98090_MIXADR_IN65DIFF_WIDTH 1 +#define M98090_MIXADR_IN34DIFF_MASK (1<<1) +#define M98090_MIXADR_IN34DIFF_SHIFT 1 +#define M98090_MIXADR_IN34DIFF_WIDTH 1 +#define M98090_MIXADR_IN12DIFF_MASK (1<<0) +#define M98090_MIXADR_IN12DIFF_SHIFT 0 +#define M98090_MIXADR_IN12DIFF_WIDTH 1 +#define M98090_MIXADR_MASK (255<<0) +#define M98090_MIXADR_SHIFT 0 +#define M98090_MIXADR_WIDTH 8 + +/* + * M98090_REG_LEFT_ADC_LEVEL + */ +#define M98090_AVLG_MASK (7<<4) +#define M98090_AVLG_SHIFT 4 +#define M98090_AVLG_WIDTH 3 +#define M98090_AVLG_NUM (1<<M98090_AVLG_WIDTH) +#define M98090_AVL_MASK (15<<0) +#define M98090_AVL_SHIFT 0 +#define M98090_AVL_WIDTH 4 +#define M98090_AVL_NUM (1<<M98090_AVL_WIDTH) + +/* + * M98090_REG_RIGHT_ADC_LEVEL + */ +#define M98090_AVRG_MASK (7<<4) +#define M98090_AVRG_SHIFT 4 +#define M98090_AVRG_WIDTH 3 +#define M98090_AVRG_NUM (1<<M98090_AVRG_WIDTH) +#define M98090_AVR_MASK (15<<0) +#define M98090_AVR_SHIFT 0 +#define M98090_AVR_WIDTH 4 +#define M98090_AVR_NUM (1<<M98090_AVR_WIDTH) + +/* + * M98090_REG_ADC_BIQUAD_LEVEL + */ +#define M98090_AVBQ_MASK (15<<0) +#define M98090_AVBQ_SHIFT 0 +#define M98090_AVBQ_WIDTH 4 +#define M98090_AVBQ_NUM (1<<M98090_AVBQ_WIDTH) + +/* + * M98090_REG_ADC_SIDETONE + */ +#define M98090_DSTSR_MASK (1<<7) +#define M98090_DSTSR_SHIFT 7 +#define M98090_DSTSR_WIDTH 1 +#define M98090_DSTSL_MASK (1<<6) +#define M98090_DSTSL_SHIFT 6 +#define M98090_DSTSL_WIDTH 1 +#define M98090_DVST_MASK (31<<0) +#define M98090_DVST_SHIFT 0 +#define M98090_DVST_WIDTH 5 +#define M98090_DVST_NUM 31 + +/* + * M98090_REG_SYSTEM_CLOCK + */ +#define M98090_PSCLK_MASK (3<<4) +#define M98090_PSCLK_SHIFT 4 +#define M98090_PSCLK_WIDTH 2 +#define M98090_PSCLK_DISABLED (0<<4) +#define M98090_PSCLK_DIV1 (1<<4) +#define M98090_PSCLK_DIV2 (2<<4) +#define M98090_PSCLK_DIV4 (3<<4) + +/* + * M98090_REG_CLOCK_MODE + */ +#define M98090_FREQ_MASK (15<<4) +#define M98090_FREQ_SHIFT 4 +#define M98090_FREQ_WIDTH 4 +#define M98090_USE_M1_MASK (1<<0) +#define M98090_USE_M1_SHIFT 0 +#define M98090_USE_M1_WIDTH 1 +#define M98090_USE_M1_NUM (1<<M98090_USE_M1_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_NI_MSB + */ +#define M98090_NI_HI_MASK (127<<0) +#define M98090_NI_HI_SHIFT 0 +#define M98090_NI_HI_WIDTH 7 +#define M98090_NI_HI_NUM (1<<M98090_NI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_NI_LSB + */ +#define M98090_NI_LO_MASK (255<<0) +#define M98090_NI_LO_SHIFT 0 +#define M98090_NI_LO_WIDTH 8 +#define M98090_NI_LO_NUM (1<<M98090_NI_LO_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_MI_MSB + */ +#define M98090_MI_HI_MASK (255<<0) +#define M98090_MI_HI_SHIFT 0 +#define M98090_MI_HI_WIDTH 8 +#define M98090_MI_HI_NUM (1<<M98090_MI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_RATIO_MI_LSB + */ +#define M98090_MI_LO_MASK (255<<0) +#define M98090_MI_LO_SHIFT 0 +#define M98090_MI_LO_WIDTH 8 +#define M98090_MI_LO_NUM (1<<M98090_MI_LO_WIDTH) + +/* + * M98090_REG_MASTER_MODE + */ +#define M98090_MAS_MASK (1<<7) +#define M98090_MAS_SHIFT 7 +#define M98090_MAS_WIDTH 1 +#define M98090_BSEL_MASK (1<<0) +#define M98090_BSEL_SHIFT 0 +#define M98090_BSEL_WIDTH 1 +#define M98090_BSEL_32 (1<<0) +#define M98090_BSEL_48 (2<<0) +#define M98090_BSEL_64 (3<<0) + +/* + * M98090_REG_INTERFACE_FORMAT + */ +#define M98090_RJ_MASK (1<<5) +#define M98090_RJ_SHIFT 5 +#define M98090_RJ_WIDTH 1 +#define M98090_WCI_MASK (1<<4) +#define M98090_WCI_SHIFT 4 +#define M98090_WCI_WIDTH 1 +#define M98090_BCI_MASK (1<<3) +#define M98090_BCI_SHIFT 3 +#define M98090_BCI_WIDTH 1 +#define M98090_DLY_MASK (1<<2) +#define M98090_DLY_SHIFT 2 +#define M98090_DLY_WIDTH 1 +#define M98090_WS_MASK (3<<0) +#define M98090_WS_SHIFT 0 +#define M98090_WS_WIDTH 2 +#define M98090_WS_NUM (1<<M98090_WS_WIDTH) + +/* + * M98090_REG_TDM_CONTROL + */ +#define M98090_FSW_MASK (1<<1) +#define M98090_FSW_SHIFT 1 +#define M98090_FSW_WIDTH 1 +#define M98090_TDM_MASK (1<<0) +#define M98090_TDM_SHIFT 0 +#define M98090_TDM_WIDTH 1 +#define M98090_TDM_NUM (1<<M98090_TDM_WIDTH) + +/* + * M98090_REG_TDM_FORMAT + */ +#define M98090_TDM_SLOTL_MASK (3<<6) +#define M98090_TDM_SLOTL_SHIFT 6 +#define M98090_TDM_SLOTL_WIDTH 2 +#define M98090_TDM_SLOTL_NUM (1<<M98090_TDM_SLOTL_WIDTH) +#define M98090_TDM_SLOTR_MASK (3<<4) +#define M98090_TDM_SLOTR_SHIFT 4 +#define M98090_TDM_SLOTR_WIDTH 2 +#define M98090_TDM_SLOTR_NUM (1<<M98090_TDM_SLOTR_WIDTH) +#define M98090_TDM_SLOTDLY_MASK (15<<0) +#define M98090_TDM_SLOTDLY_SHIFT 0 +#define M98090_TDM_SLOTDLY_WIDTH 4 +#define M98090_TDM_SLOTDLY_NUM (1<<M98090_TDM_SLOTDLY_WIDTH) + +/* + * M98090_REG_IO_CONFIGURATION + */ +#define M98090_LTEN_MASK (1<<5) +#define M98090_LTEN_SHIFT 5 +#define M98090_LTEN_WIDTH 1 +#define M98090_LTEN_NUM (1<<M98090_LTEN_WIDTH) +#define M98090_LBEN_MASK (1<<4) +#define M98090_LBEN_SHIFT 4 +#define M98090_LBEN_WIDTH 1 +#define M98090_LBEN_NUM (1<<M98090_LBEN_WIDTH) +#define M98090_DMONO_MASK (1<<3) +#define M98090_DMONO_SHIFT 3 +#define M98090_DMONO_WIDTH 1 +#define M98090_DMONO_NUM (1<<M98090_DMONO_WIDTH) +#define M98090_HIZOFF_MASK (1<<2) +#define M98090_HIZOFF_SHIFT 2 +#define M98090_HIZOFF_WIDTH 1 +#define M98090_HIZOFF_NUM (1<<M98090_HIZOFF_WIDTH) +#define M98090_SDOEN_MASK (1<<1) +#define M98090_SDOEN_SHIFT 1 +#define M98090_SDOEN_WIDTH 1 +#define M98090_SDOEN_NUM (1<<M98090_SDOEN_WIDTH) +#define M98090_SDIEN_MASK (1<<0) +#define M98090_SDIEN_SHIFT 0 +#define M98090_SDIEN_WIDTH 1 +#define M98090_SDIEN_NUM (1<<M98090_SDIEN_WIDTH) + +/* + * M98090_REG_FILTER_CONFIG + */ +#define M98090_MODE_MASK (1<<7) +#define M98090_MODE_SHIFT 7 +#define M98090_MODE_WIDTH 1 +#define M98090_AHPF_MASK (1<<6) +#define M98090_AHPF_SHIFT 6 +#define M98090_AHPF_WIDTH 1 +#define M98090_AHPF_NUM (1<<M98090_AHPF_WIDTH) +#define M98090_DHPF_MASK (1<<5) +#define M98090_DHPF_SHIFT 5 +#define M98090_DHPF_WIDTH 1 +#define M98090_DHPF_NUM (1<<M98090_DHPF_WIDTH) +#define M98090_DHF_MASK (1<<4) +#define M98090_DHF_SHIFT 4 +#define M98090_DHF_WIDTH 1 +#define M98090_FLT_DMIC34MODE_MASK (1<<3) +#define M98090_FLT_DMIC34MODE_SHIFT 3 +#define M98090_FLT_DMIC34MODE_WIDTH 1 +#define M98090_FLT_DMIC34HPF_MASK (1<<2) +#define M98090_FLT_DMIC34HPF_SHIFT 2 +#define M98090_FLT_DMIC34HPF_WIDTH 1 +#define M98090_FLT_DMIC34HPF_NUM (1<<M98090_FLT_DMIC34HPF_WIDTH) + +/* + * M98090_REG_DAI_PLAYBACK_LEVEL + */ +#define M98090_DVM_MASK (1<<7) +#define M98090_DVM_SHIFT 7 +#define M98090_DVM_WIDTH 1 +#define M98090_DVG_MASK (3<<4) +#define M98090_DVG_SHIFT 4 +#define M98090_DVG_WIDTH 2 +#define M98090_DVG_NUM (1<<M98090_DVG_WIDTH) +#define M98090_DV_MASK (15<<0) +#define M98090_DV_SHIFT 0 +#define M98090_DV_WIDTH 4 +#define M98090_DV_NUM (1<<M98090_DV_WIDTH) + +/* + * M98090_REG_DAI_PLAYBACK_LEVEL_EQ + */ +#define M98090_EQCLPN_MASK (1<<4) +#define M98090_EQCLPN_SHIFT 4 +#define M98090_EQCLPN_WIDTH 1 +#define M98090_EQCLPN_NUM (1<<M98090_EQCLPN_WIDTH) +#define M98090_DVEQ_MASK (15<<0) +#define M98090_DVEQ_SHIFT 0 +#define M98090_DVEQ_WIDTH 4 +#define M98090_DVEQ_NUM (1<<M98090_DVEQ_WIDTH) + +/* + * M98090_REG_LEFT_HP_MIXER + */ +#define M98090_MIXHPL_MIC2_MASK (1<<5) +#define M98090_MIXHPL_MIC2_SHIFT 5 +#define M98090_MIXHPL_MIC2_WIDTH 1 +#define M98090_MIXHPL_MIC1_MASK (1<<4) +#define M98090_MIXHPL_MIC1_SHIFT 4 +#define M98090_MIXHPL_MIC1_WIDTH 1 +#define M98090_MIXHPL_LINEB_MASK (1<<3) +#define M98090_MIXHPL_LINEB_SHIFT 3 +#define M98090_MIXHPL_LINEB_WIDTH 1 +#define M98090_MIXHPL_LINEA_MASK (1<<2) +#define M98090_MIXHPL_LINEA_SHIFT 2 +#define M98090_MIXHPL_LINEA_WIDTH 1 +#define M98090_MIXHPL_DACR_MASK (1<<1) +#define M98090_MIXHPL_DACR_SHIFT 1 +#define M98090_MIXHPL_DACR_WIDTH 1 +#define M98090_MIXHPL_DACL_MASK (1<<0) +#define M98090_MIXHPL_DACL_SHIFT 0 +#define M98090_MIXHPL_DACL_WIDTH 1 +#define M98090_MIXHPL_MASK (63<<0) +#define M98090_MIXHPL_SHIFT 0 +#define M98090_MIXHPL_WIDTH 6 + +/* + * M98090_REG_RIGHT_HP_MIXER + */ +#define M98090_MIXHPR_MIC2_MASK (1<<5) +#define M98090_MIXHPR_MIC2_SHIFT 5 +#define M98090_MIXHPR_MIC2_WIDTH 1 +#define M98090_MIXHPR_MIC1_MASK (1<<4) +#define M98090_MIXHPR_MIC1_SHIFT 4 +#define M98090_MIXHPR_MIC1_WIDTH 1 +#define M98090_MIXHPR_LINEB_MASK (1<<3) +#define M98090_MIXHPR_LINEB_SHIFT 3 +#define M98090_MIXHPR_LINEB_WIDTH 1 +#define M98090_MIXHPR_LINEA_MASK (1<<2) +#define M98090_MIXHPR_LINEA_SHIFT 2 +#define M98090_MIXHPR_LINEA_WIDTH 1 +#define M98090_MIXHPR_DACR_MASK (1<<1) +#define M98090_MIXHPR_DACR_SHIFT 1 +#define M98090_MIXHPR_DACR_WIDTH 1 +#define M98090_MIXHPR_DACL_MASK (1<<0) +#define M98090_MIXHPR_DACL_SHIFT 0 +#define M98090_MIXHPR_DACL_WIDTH 1 +#define M98090_MIXHPR_MASK (63<<0) +#define M98090_MIXHPR_SHIFT 0 +#define M98090_MIXHPR_WIDTH 6 + +/* + * M98090_REG_HP_CONTROL + */ +#define M98090_MIXHPRSEL_MASK (1<<5) +#define M98090_MIXHPRSEL_SHIFT 5 +#define M98090_MIXHPRSEL_WIDTH 1 +#define M98090_MIXHPLSEL_MASK (1<<4) +#define M98090_MIXHPLSEL_SHIFT 4 +#define M98090_MIXHPLSEL_WIDTH 1 +#define M98090_MIXHPRG_MASK (3<<2) +#define M98090_MIXHPRG_SHIFT 2 +#define M98090_MIXHPRG_WIDTH 2 +#define M98090_MIXHPRG_NUM (1<<M98090_MIXHPRG_WIDTH) +#define M98090_MIXHPLG_MASK (3<<0) +#define M98090_MIXHPLG_SHIFT 0 +#define M98090_MIXHPLG_WIDTH 2 +#define M98090_MIXHPLG_NUM (1<<M98090_MIXHPLG_WIDTH) + +/* + * M98090_REG_LEFT_HP_VOLUME + */ +#define M98090_HPLM_MASK (1<<7) +#define M98090_HPLM_SHIFT 7 +#define M98090_HPLM_WIDTH 1 +#define M98090_HPVOLL_MASK (31<<0) +#define M98090_HPVOLL_SHIFT 0 +#define M98090_HPVOLL_WIDTH 5 +#define M98090_HPVOLL_NUM (1<<M98090_HPVOLL_WIDTH) + +/* + * M98090_REG_RIGHT_HP_VOLUME + */ +#define M98090_HPRM_MASK (1<<7) +#define M98090_HPRM_SHIFT 7 +#define M98090_HPRM_WIDTH 1 +#define M98090_HPVOLR_MASK (31<<0) +#define M98090_HPVOLR_SHIFT 0 +#define M98090_HPVOLR_WIDTH 5 +#define M98090_HPVOLR_NUM (1<<M98090_HPVOLR_WIDTH) + +/* + * M98090_REG_LEFT_SPK_MIXER + */ +#define M98090_MIXSPL_MIC2_MASK (1<<5) +#define M98090_MIXSPL_MIC2_SHIFT 5 +#define M98090_MIXSPL_MIC2_WIDTH 1 +#define M98090_MIXSPL_MIC1_MASK (1<<4) +#define M98090_MIXSPL_MIC1_SHIFT 4 +#define M98090_MIXSPL_MIC1_WIDTH 1 +#define M98090_MIXSPL_LINEB_MASK (1<<3) +#define M98090_MIXSPL_LINEB_SHIFT 3 +#define M98090_MIXSPL_LINEB_WIDTH 1 +#define M98090_MIXSPL_LINEA_MASK (1<<2) +#define M98090_MIXSPL_LINEA_SHIFT 2 +#define M98090_MIXSPL_LINEA_WIDTH 1 +#define M98090_MIXSPL_DACR_MASK (1<<1) +#define M98090_MIXSPL_DACR_SHIFT 1 +#define M98090_MIXSPL_DACR_WIDTH 1 +#define M98090_MIXSPL_DACL_MASK (1<<0) +#define M98090_MIXSPL_DACL_SHIFT 0 +#define M98090_MIXSPL_DACL_WIDTH 1 +#define M98090_MIXSPL_MASK (63<<0) +#define M98090_MIXSPL_SHIFT 0 +#define M98090_MIXSPL_WIDTH 6 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 + + +/* + * M98090_REG_RIGHT_SPK_MIXER + */ +#define M98090_SPK_SLAVE_MASK (1<<6) +#define M98090_SPK_SLAVE_SHIFT 6 +#define M98090_SPK_SLAVE_WIDTH 1 +#define M98090_MIXSPR_MIC2_MASK (1<<5) +#define M98090_MIXSPR_MIC2_SHIFT 5 +#define M98090_MIXSPR_MIC2_WIDTH 1 +#define M98090_MIXSPR_MIC1_MASK (1<<4) +#define M98090_MIXSPR_MIC1_SHIFT 4 +#define M98090_MIXSPR_MIC1_WIDTH 1 +#define M98090_MIXSPR_LINEB_MASK (1<<3) +#define M98090_MIXSPR_LINEB_SHIFT 3 +#define M98090_MIXSPR_LINEB_WIDTH 1 +#define M98090_MIXSPR_LINEA_MASK (1<<2) +#define M98090_MIXSPR_LINEA_SHIFT 2 +#define M98090_MIXSPR_LINEA_WIDTH 1 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 +#define M98090_MIXSPR_DACL_MASK (1<<0) +#define M98090_MIXSPR_DACL_SHIFT 0 +#define M98090_MIXSPR_DACL_WIDTH 1 +#define M98090_MIXSPR_MASK (63<<0) +#define M98090_MIXSPR_SHIFT 0 +#define M98090_MIXSPR_WIDTH 6 + +/* + * M98090_REG_SPK_CONTROL + */ +#define M98090_MIXSPRG_MASK (3<<2) +#define M98090_MIXSPRG_SHIFT 2 +#define M98090_MIXSPRG_WIDTH 2 +#define M98090_MIXSPRG_NUM (1<<M98090_MIXSPRG_WIDTH) +#define M98090_MIXSPLG_MASK (3<<0) +#define M98090_MIXSPLG_SHIFT 0 +#define M98090_MIXSPLG_WIDTH 2 +#define M98090_MIXSPLG_NUM (1<<M98090_MIXSPLG_WIDTH) + +/* + * M98090_REG_LEFT_SPK_VOLUME + */ +#define M98090_SPLM_MASK (1<<7) +#define M98090_SPLM_SHIFT 7 +#define M98090_SPLM_WIDTH 1 +#define M98090_SPVOLL_MASK (63<<0) +#define M98090_SPVOLL_SHIFT 0 +#define M98090_SPVOLL_WIDTH 6 +#define M98090_SPVOLL_NUM 40 + +/* + * M98090_REG_RIGHT_SPK_VOLUME + */ +#define M98090_SPRM_MASK (1<<7) +#define M98090_SPRM_SHIFT 7 +#define M98090_SPRM_WIDTH 1 +#define M98090_SPVOLR_MASK (63<<0) +#define M98090_SPVOLR_SHIFT 0 +#define M98090_SPVOLR_WIDTH 6 +#define M98090_SPVOLR_NUM 40 + +/* + * M98090_REG_DRC_TIMING + */ +#define M98090_DRCEN_MASK (1<<7) +#define M98090_DRCEN_SHIFT 7 +#define M98090_DRCEN_WIDTH 1 +#define M98090_DRCEN_NUM (1<<M98090_DRCEN_WIDTH) +#define M98090_DRCRLS_MASK (7<<4) +#define M98090_DRCRLS_SHIFT 4 +#define M98090_DRCRLS_WIDTH 3 +#define M98090_DRCATK_MASK (7<<0) +#define M98090_DRCATK_SHIFT 0 +#define M98090_DRCATK_WIDTH 3 + +/* + * M98090_REG_DRC_COMPRESSOR + */ +#define M98090_DRCCMP_MASK (7<<5) +#define M98090_DRCCMP_SHIFT 5 +#define M98090_DRCCMP_WIDTH 3 +#define M98090_DRCTHC_MASK (31<<0) +#define M98090_DRCTHC_SHIFT 0 +#define M98090_DRCTHC_WIDTH 5 +#define M98090_DRCTHC_NUM (1<<M98090_DRCTHC_WIDTH) + +/* + * M98090_REG_DRC_EXPANDER + */ +#define M98090_DRCEXP_MASK (7<<5) +#define M98090_DRCEXP_SHIFT 5 +#define M98090_DRCEXP_WIDTH 3 +#define M98090_DRCTHE_MASK (31<<0) +#define M98090_DRCTHE_SHIFT 0 +#define M98090_DRCTHE_WIDTH 5 +#define M98090_DRCTHE_NUM (1<<M98090_DRCTHE_WIDTH) + +/* + * M98090_REG_DRC_GAIN + */ +#define M98090_DRCG_MASK (31<<0) +#define M98090_DRCG_SHIFT 0 +#define M98090_DRCG_WIDTH 5 +#define M98090_DRCG_NUM 13 + +/* + * M98090_REG_RCV_LOUTL_MIXER + */ +#define M98090_MIXRCVL_MIC2_MASK (1<<5) +#define M98090_MIXRCVL_MIC2_SHIFT 5 +#define M98090_MIXRCVL_MIC2_WIDTH 1 +#define M98090_MIXRCVL_MIC1_MASK (1<<4) +#define M98090_MIXRCVL_MIC1_SHIFT 4 +#define M98090_MIXRCVL_MIC1_WIDTH 1 +#define M98090_MIXRCVL_LINEB_MASK (1<<3) +#define M98090_MIXRCVL_LINEB_SHIFT 3 +#define M98090_MIXRCVL_LINEB_WIDTH 1 +#define M98090_MIXRCVL_LINEA_MASK (1<<2) +#define M98090_MIXRCVL_LINEA_SHIFT 2 +#define M98090_MIXRCVL_LINEA_WIDTH 1 +#define M98090_MIXRCVL_DACR_MASK (1<<1) +#define M98090_MIXRCVL_DACR_SHIFT 1 +#define M98090_MIXRCVL_DACR_WIDTH 1 +#define M98090_MIXRCVL_DACL_MASK (1<<0) +#define M98090_MIXRCVL_DACL_SHIFT 0 +#define M98090_MIXRCVL_DACL_WIDTH 1 +#define M98090_MIXRCVL_MASK (63<<0) +#define M98090_MIXRCVL_SHIFT 0 +#define M98090_MIXRCVL_WIDTH 6 + +/* + * M98090_REG_RCV_LOUTL_CONTROL + */ +#define M98090_MIXRCVLG_MASK (3<<0) +#define M98090_MIXRCVLG_SHIFT 0 +#define M98090_MIXRCVLG_WIDTH 2 +#define M98090_MIXRCVLG_NUM (1<<M98090_MIXRCVLG_WIDTH) + +/* + * M98090_REG_RCV_LOUTL_VOLUME + */ +#define M98090_RCVLM_MASK (1<<7) +#define M98090_RCVLM_SHIFT 7 +#define M98090_RCVLM_WIDTH 1 +#define M98090_RCVLVOL_MASK (31<<0) +#define M98090_RCVLVOL_SHIFT 0 +#define M98090_RCVLVOL_WIDTH 5 +#define M98090_RCVLVOL_NUM (1<<M98090_RCVLVOL_WIDTH) + +/* + * M98090_REG_LOUTR_MIXER + */ +#define M98090_LINMOD_MASK (1<<7) +#define M98090_LINMOD_SHIFT 7 +#define M98090_LINMOD_WIDTH 1 +#define M98090_MIXRCVR_MIC2_MASK (1<<5) +#define M98090_MIXRCVR_MIC2_SHIFT 5 +#define M98090_MIXRCVR_MIC2_WIDTH 1 +#define M98090_MIXRCVR_MIC1_MASK (1<<4) +#define M98090_MIXRCVR_MIC1_SHIFT 4 +#define M98090_MIXRCVR_MIC1_WIDTH 1 +#define M98090_MIXRCVR_LINEB_MASK (1<<3) +#define M98090_MIXRCVR_LINEB_SHIFT 3 +#define M98090_MIXRCVR_LINEB_WIDTH 1 +#define M98090_MIXRCVR_LINEA_MASK (1<<2) +#define M98090_MIXRCVR_LINEA_SHIFT 2 +#define M98090_MIXRCVR_LINEA_WIDTH 1 +#define M98090_MIXRCVR_DACR_MASK (1<<1) +#define M98090_MIXRCVR_DACR_SHIFT 1 +#define M98090_MIXRCVR_DACR_WIDTH 1 +#define M98090_MIXRCVR_DACL_MASK (1<<0) +#define M98090_MIXRCVR_DACL_SHIFT 0 +#define M98090_MIXRCVR_DACL_WIDTH 1 +#define M98090_MIXRCVR_MASK (63<<0) +#define M98090_MIXRCVR_SHIFT 0 +#define M98090_MIXRCVR_WIDTH 6 + +/* + * M98090_REG_LOUTR_CONTROL + */ +#define M98090_MIXRCVRG_MASK (3<<0) +#define M98090_MIXRCVRG_SHIFT 0 +#define M98090_MIXRCVRG_WIDTH 2 +#define M98090_MIXRCVRG_NUM (1<<M98090_MIXRCVRG_WIDTH) + +/* + * M98090_REG_LOUTR_VOLUME + */ +#define M98090_RCVRM_MASK (1<<7) +#define M98090_RCVRM_SHIFT 7 +#define M98090_RCVRM_WIDTH 1 +#define M98090_RCVRVOL_MASK (31<<0) +#define M98090_RCVRVOL_SHIFT 0 +#define M98090_RCVRVOL_WIDTH 5 +#define M98090_RCVRVOL_NUM (1<<M98090_RCVRVOL_WIDTH) + +/* + * M98090_REG_JACK_DETECT + */ +#define M98090_JDETEN_MASK (1<<7) +#define M98090_JDETEN_SHIFT 7 +#define M98090_JDETEN_WIDTH 1 +#define M98090_JDWK_MASK (1<<6) +#define M98090_JDWK_SHIFT 6 +#define M98090_JDWK_WIDTH 1 +#define M98090_JDEB_MASK (3<<0) +#define M98090_JDEB_SHIFT 0 +#define M98090_JDEB_WIDTH 2 +#define M98090_JDEB_25MS (0<<0) +#define M98090_JDEB_50MS (1<<0) +#define M98090_JDEB_100MS (2<<0) +#define M98090_JDEB_200MS (3<<0) + +/* + * M98090_REG_INPUT_ENABLE + */ +#define M98090_MBEN_MASK (1<<4) +#define M98090_MBEN_SHIFT 4 +#define M98090_MBEN_WIDTH 1 +#define M98090_LINEAEN_MASK (1<<3) +#define M98090_LINEAEN_SHIFT 3 +#define M98090_LINEAEN_WIDTH 1 +#define M98090_LINEBEN_MASK (1<<2) +#define M98090_LINEBEN_SHIFT 2 +#define M98090_LINEBEN_WIDTH 1 +#define M98090_ADREN_MASK (1<<1) +#define M98090_ADREN_SHIFT 1 +#define M98090_ADREN_WIDTH 1 +#define M98090_ADLEN_MASK (1<<0) +#define M98090_ADLEN_SHIFT 0 +#define M98090_ADLEN_WIDTH 1 + +/* + * M98090_REG_OUTPUT_ENABLE + */ +#define M98090_HPREN_MASK (1<<7) +#define M98090_HPREN_SHIFT 7 +#define M98090_HPREN_WIDTH 1 +#define M98090_HPLEN_MASK (1<<6) +#define M98090_HPLEN_SHIFT 6 +#define M98090_HPLEN_WIDTH 1 +#define M98090_SPREN_MASK (1<<5) +#define M98090_SPREN_SHIFT 5 +#define M98090_SPREN_WIDTH 1 +#define M98090_SPLEN_MASK (1<<4) +#define M98090_SPLEN_SHIFT 4 +#define M98090_SPLEN_WIDTH 1 +#define M98090_RCVLEN_MASK (1<<3) +#define M98090_RCVLEN_SHIFT 3 +#define M98090_RCVLEN_WIDTH 1 +#define M98090_RCVREN_MASK (1<<2) +#define M98090_RCVREN_SHIFT 2 +#define M98090_RCVREN_WIDTH 1 +#define M98090_DAREN_MASK (1<<1) +#define M98090_DAREN_SHIFT 1 +#define M98090_DAREN_WIDTH 1 +#define M98090_DALEN_MASK (1<<0) +#define M98090_DALEN_SHIFT 0 +#define M98090_DALEN_WIDTH 1 + +/* + * M98090_REG_LEVEL_CONTROL + */ +#define M98090_ZDENN_MASK (1<<2) +#define M98090_ZDENN_SHIFT 2 +#define M98090_ZDENN_WIDTH 1 +#define M98090_ZDENN_NUM (1<<M98090_ZDENN_WIDTH) +#define M98090_VS2ENN_MASK (1<<1) +#define M98090_VS2ENN_SHIFT 1 +#define M98090_VS2ENN_WIDTH 1 +#define M98090_VS2ENN_NUM (1<<M98090_VS2ENN_WIDTH) +#define M98090_VSENN_MASK (1<<0) +#define M98090_VSENN_SHIFT 0 +#define M98090_VSENN_WIDTH 1 +#define M98090_VSENN_NUM (1<<M98090_VSENN_WIDTH) + +/* + * M98090_REG_DSP_FILTER_ENABLE + */ +#define M98090_DMIC34BQEN_MASK (1<<4) +#define M98090_DMIC34BQEN_SHIFT 4 +#define M98090_DMIC34BQEN_WIDTH 1 +#define M98090_DMIC34BQEN_NUM (1<<M98090_DMIC34BQEN_WIDTH) +#define M98090_ADCBQEN_MASK (1<<3) +#define M98090_ADCBQEN_SHIFT 3 +#define M98090_ADCBQEN_WIDTH 1 +#define M98090_ADCBQEN_NUM (1<<M98090_ADCBQEN_WIDTH) +#define M98090_EQ3BANDEN_MASK (1<<2) +#define M98090_EQ3BANDEN_SHIFT 2 +#define M98090_EQ3BANDEN_WIDTH 1 +#define M98090_EQ3BANDEN_NUM (1<<M98090_EQ3BANDEN_WIDTH) +#define M98090_EQ5BANDEN_MASK (1<<1) +#define M98090_EQ5BANDEN_SHIFT 1 +#define M98090_EQ5BANDEN_WIDTH 1 +#define M98090_EQ5BANDEN_NUM (1<<M98090_EQ5BANDEN_WIDTH) +#define M98090_EQ7BANDEN_MASK (1<<0) +#define M98090_EQ7BANDEN_SHIFT 0 +#define M98090_EQ7BANDEN_WIDTH 1 +#define M98090_EQ7BANDEN_NUM (1<<M98090_EQ7BANDEN_WIDTH) + +/* + * M98090_REG_BIAS_CONTROL + */ +#define M98090_VCM_MODE_MASK (1<<0) +#define M98090_VCM_MODE_SHIFT 0 +#define M98090_VCM_MODE_WIDTH 1 +#define M98090_VCM_MODE_NUM (1<<M98090_VCM_MODE_WIDTH) + +/* + * M98090_REG_DAC_CONTROL + */ +#define M98090_PERFMODE_MASK (1<<1) +#define M98090_PERFMODE_SHIFT 1 +#define M98090_PERFMODE_WIDTH 1 +#define M98090_PERFMODE_NUM (1<<M98090_PERFMODE_WIDTH) +#define M98090_DACHP_MASK (1<<0) +#define M98090_DACHP_SHIFT 0 +#define M98090_DACHP_WIDTH 1 +#define M98090_DACHP_NUM (1<<M98090_DACHP_WIDTH) + +/* + * M98090_REG_ADC_CONTROL + */ +#define M98090_OSR128_MASK (1<<2) +#define M98090_OSR128_SHIFT 2 +#define M98090_OSR128_WIDTH 1 +#define M98090_ADCDITHER_MASK (1<<1) +#define M98090_ADCDITHER_SHIFT 1 +#define M98090_ADCDITHER_WIDTH 1 +#define M98090_ADCDITHER_NUM (1<<M98090_ADCDITHER_WIDTH) +#define M98090_ADCHP_MASK (1<<0) +#define M98090_ADCHP_SHIFT 0 +#define M98090_ADCHP_WIDTH 1 +#define M98090_ADCHP_NUM (1<<M98090_ADCHP_WIDTH) + +/* + * M98090_REG_DEVICE_SHUTDOWN + */ +#define M98090_SHDNN_MASK (1<<7) +#define M98090_SHDNN_SHIFT 7 +#define M98090_SHDNN_WIDTH 1 + +/* + * M98090_REG_EQUALIZER_BASE + */ +#define M98090_B0_1_HI_MASK (255<<0) +#define M98090_B0_1_HI_SHIFT 0 +#define M98090_B0_1_HI_WIDTH 8 +#define M98090_B0_1_MID_MASK (255<<0) +#define M98090_B0_1_MID_SHIFT 0 +#define M98090_B0_1_MID_WIDTH 8 +#define M98090_B0_1_LO_MASK (255<<0) +#define M98090_B0_1_LO_SHIFT 0 +#define M98090_B0_1_LO_WIDTH 8 +#define M98090_B1_1_HI_MASK (255<<0) +#define M98090_B1_1_HI_SHIFT 0 +#define M98090_B1_1_HI_WIDTH 8 +#define M98090_B1_1_MID_MASK (255<<0) +#define M98090_B1_1_MID_SHIFT 0 +#define M98090_B1_1_MID_WIDTH 8 +#define M98090_B1_1_LO_MASK (255<<0) +#define M98090_B1_1_LO_SHIFT 0 +#define M98090_B1_1_LO_WIDTH 8 +#define M98090_B2_1_HI_MASK (255<<0) +#define M98090_B2_1_HI_SHIFT 0 +#define M98090_B2_1_HI_WIDTH 8 +#define M98090_B2_1_MID_MASK (255<<0) +#define M98090_B2_1_MID_SHIFT 0 +#define M98090_B2_1_MID_WIDTH 8 +#define M98090_B2_1_LO_MASK (255<<0) +#define M98090_B2_1_LO_SHIFT 0 +#define M98090_B2_1_LO_WIDTH 8 +#define M98090_A1_1_HI_MASK (255<<0) +#define M98090_A1_1_HI_SHIFT 0 +#define M98090_A1_1_HI_WIDTH 8 +#define M98090_A1_1_MID_MASK (255<<0) +#define M98090_A1_1_MID_SHIFT 0 +#define M98090_A1_1_MID_WIDTH 8 +#define M98090_A1_1_LO_MASK (255<<0) +#define M98090_A1_1_LO_SHIFT 0 +#define M98090_A1_1_LO_WIDTH 8 +#define M98090_A2_1_HI_MASK (255<<0) +#define M98090_A2_1_HI_SHIFT 0 +#define M98090_A2_1_HI_WIDTH 8 +#define M98090_A2_1_MID_MASK (255<<0) +#define M98090_A2_1_MID_SHIFT 0 +#define M98090_A2_1_MID_WIDTH 8 +#define M98090_A2_1_LO_MASK (255<<0) +#define M98090_A2_1_LO_SHIFT 0 +#define M98090_A2_1_LO_WIDTH 8 + +#define M98090_COEFS_PER_BAND 5 +#define M98090_COEFS_BLK_SZ (M98090_COEFS_PER_BAND * 3) +#define M98090_COEFS_MAX_SZ (M98090_COEFS_BLK_SZ * 7) + +/* + * M98090_REG_RECORD_BIQUAD_BASE + */ +#define M98090_REC_B0_HI_MASK (255<<0) +#define M98090_REC_B0_HI_SHIFT 0 +#define M98090_REC_B0_HI_WIDTH 8 +#define M98090_REC_B0_MID_MASK (255<<0) +#define M98090_REC_B0_MID_SHIFT 0 +#define M98090_REC_B0_MID_WIDTH 8 +#define M98090_REC_B0_LO_MASK (255<<0) +#define M98090_REC_B0_LO_SHIFT 0 +#define M98090_REC_B0_LO_WIDTH 8 +#define M98090_REC_B1_HI_MASK (255<<0) +#define M98090_REC_B1_HI_SHIFT 0 +#define M98090_REC_B1_HI_WIDTH 8 +#define M98090_REC_B1_MID_MASK (255<<0) +#define M98090_REC_B1_MID_SHIFT 0 +#define M98090_REC_B1_MID_WIDTH 8 +#define M98090_REC_B1_LO_MASK (255<<0) +#define M98090_REC_B1_LO_SHIFT 0 +#define M98090_REC_B1_LO_WIDTH 8 +#define M98090_REC_B2_HI_MASK (255<<0) +#define M98090_REC_B2_HI_SHIFT 0 +#define M98090_REC_B2_HI_WIDTH 8 +#define M98090_REC_B2_MID_MASK (255<<0) +#define M98090_REC_B2_MID_SHIFT 0 +#define M98090_REC_B2_MID_WIDTH 8 +#define M98090_REC_B2_LO_MASK (255<<0) +#define M98090_REC_B2_LO_SHIFT 0 +#define M98090_REC_B2_LO_WIDTH 8 +#define M98090_REC_A1_HI_MASK (255<<0) +#define M98090_REC_A1_HI_SHIFT 0 +#define M98090_REC_A1_HI_WIDTH 8 +#define M98090_REC_A1_MID_MASK (255<<0) +#define M98090_REC_A1_MID_SHIFT 0 +#define M98090_REC_A1_MID_WIDTH 8 +#define M98090_REC_A1_LO_MASK (255<<0) +#define M98090_REC_A1_LO_SHIFT 0 +#define M98090_REC_A1_LO_WIDTH 8 +#define M98090_REC_A2_HI_MASK (255<<0) +#define M98090_REC_A2_HI_SHIFT 0 +#define M98090_REC_A2_HI_WIDTH 8 +#define M98090_REC_A2_MID_MASK (255<<0) +#define M98090_REC_A2_MID_SHIFT 0 +#define M98090_REC_A2_MID_WIDTH 8 +#define M98090_REC_A2_LO_MASK (255<<0) +#define M98090_REC_A2_LO_SHIFT 0 +#define M98090_REC_A2_LO_WIDTH 8 + +/* + * M98090_REG_DMIC3_VOLUME + */ +#define M98090_DMIC_AV3G_MASK (7<<4) +#define M98090_DMIC_AV3G_SHIFT 4 +#define M98090_DMIC_AV3G_WIDTH 3 +#define M98090_DMIC_AV3G_NUM (1<<M98090_DMIC_AV3G_WIDTH) +#define M98090_DMIC_AV3_MASK (15<<0) +#define M98090_DMIC_AV3_SHIFT 0 +#define M98090_DMIC_AV3_WIDTH 4 +#define M98090_DMIC_AV3_NUM (1<<M98090_DMIC_AV3_WIDTH) + +/* + * M98090_REG_DMIC4_VOLUME + */ +#define M98090_DMIC_AV4G_MASK (7<<4) +#define M98090_DMIC_AV4G_SHIFT 4 +#define M98090_DMIC_AV4G_WIDTH 3 +#define M98090_DMIC_AV4G_NUM (1<<M98090_DMIC_AV4G_WIDTH) +#define M98090_DMIC_AV4_MASK (15<<0) +#define M98090_DMIC_AV4_SHIFT 0 +#define M98090_DMIC_AV4_WIDTH 4 +#define M98090_DMIC_AV4_NUM (1<<M98090_DMIC_AV4_WIDTH) + +/* + * M98090_REG_DMIC34_BQ_PREATTEN + */ +#define M98090_AV34BQ_MASK (15<<0) +#define M98090_AV34BQ_SHIFT 0 +#define M98090_AV34BQ_WIDTH 4 +#define M98090_AV34BQ_NUM (1<<M98090_AV34BQ_WIDTH) + +/* + * M98090_REG_RECORD_TDM_SLOT + */ +#define M98090_TDM_SLOTADCL_MASK (3<<6) +#define M98090_TDM_SLOTADCL_SHIFT 6 +#define M98090_TDM_SLOTADCL_WIDTH 2 +#define M98090_TDM_SLOTADCL_NUM (1<<M98090_TDM_SLOTADCL_WIDTH) +#define M98090_TDM_SLOTADCR_MASK (3<<4) +#define M98090_TDM_SLOTADCR_SHIFT 4 +#define M98090_TDM_SLOTADCR_WIDTH 2 +#define M98090_TDM_SLOTADCR_NUM (1<<M98090_TDM_SLOTADCR_WIDTH) +#define M98090_TDM_SLOTDMIC3_MASK (3<<2) +#define M98090_TDM_SLOTDMIC3_SHIFT 2 +#define M98090_TDM_SLOTDMIC3_WIDTH 2 +#define M98090_TDM_SLOTDMIC3_NUM (1<<M98090_TDM_SLOTDMIC3_WIDTH) +#define M98090_TDM_SLOTDMIC4_MASK (3<<0) +#define M98090_TDM_SLOTDMIC4_SHIFT 0 +#define M98090_TDM_SLOTDMIC4_WIDTH 2 +#define M98090_TDM_SLOTDMIC4_NUM (1<<M98090_TDM_SLOTDMIC4_WIDTH) + +/* + * M98090_REG_SAMPLE_RATE + */ +#define M98090_DMIC34_ZEROPAD_MASK (1<<4) +#define M98090_DMIC34_ZEROPAD_SHIFT 4 +#define M98090_DMIC34_ZEROPAD_WIDTH 1 +#define M98090_DMIC34_ZEROPAD_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DMIC34_SRDIV_MASK (7<<0) +#define M98090_DMIC34_SRDIV_SHIFT 0 +#define M98090_DMIC34_SRDIV_WIDTH 3 + +/* + * M98090_REG_DMIC34_BIQUAD_BASE + */ +#define M98090_DMIC34_B0_HI_MASK (255<<0) +#define M98090_DMIC34_B0_HI_SHIFT 0 +#define M98090_DMIC34_B0_HI_WIDTH 8 +#define M98090_DMIC34_B0_MID_MASK (255<<0) +#define M98090_DMIC34_B0_MID_SHIFT 0 +#define M98090_DMIC34_B0_MID_WIDTH 8 +#define M98090_DMIC34_B0_LO_MASK (255<<0) +#define M98090_DMIC34_B0_LO_SHIFT 0 +#define M98090_DMIC34_B0_LO_WIDTH 8 +#define M98090_DMIC34_B1_HI_MASK (255<<0) +#define M98090_DMIC34_B1_HI_SHIFT 0 +#define M98090_DMIC34_B1_HI_WIDTH 8 +#define M98090_DMIC34_B1_MID_MASK (255<<0) +#define M98090_DMIC34_B1_MID_SHIFT 0 +#define M98090_DMIC34_B1_MID_WIDTH 8 +#define M98090_DMIC34_B1_LO_MASK (255<<0) +#define M98090_DMIC34_B1_LO_SHIFT 0 +#define M98090_DMIC34_B1_LO_WIDTH 8 +#define M98090_DMIC34_B2_HI_MASK (255<<0) +#define M98090_DMIC34_B2_HI_SHIFT 0 +#define M98090_DMIC34_B2_HI_WIDTH 8 +#define M98090_DMIC34_B2_MID_MASK (255<<0) +#define M98090_DMIC34_B2_MID_SHIFT 0 +#define M98090_DMIC34_B2_MID_WIDTH 8 +#define M98090_DMIC34_B2_LO_MASK (255<<0) +#define M98090_DMIC34_B2_LO_SHIFT 0 +#define M98090_DMIC34_B2_LO_WIDTH 8 +#define M98090_DMIC34_A1_HI_MASK (255<<0) +#define M98090_DMIC34_A1_HI_SHIFT 0 +#define M98090_DMIC34_A1_HI_WIDTH 8 +#define M98090_DMIC34_A1_MID_MASK (255<<0) +#define M98090_DMIC34_A1_MID_SHIFT 0 +#define M98090_DMIC34_A1_MID_WIDTH 8 +#define M98090_DMIC34_A1_LO_MASK (255<<0) +#define M98090_DMIC34_A1_LO_SHIFT 0 +#define M98090_DMIC34_A1_LO_WIDTH 8 +#define M98090_DMIC34_A2_HI_MASK (255<<0) +#define M98090_DMIC34_A2_HI_SHIFT 0 +#define M98090_DMIC34_A2_HI_WIDTH 8 +#define M98090_DMIC34_A2_MID_MASK (255<<0) +#define M98090_DMIC34_A2_MID_SHIFT 0 +#define M98090_DMIC34_A2_MID_WIDTH 8 +#define M98090_DMIC34_A2_LO_MASK (255<<0) +#define M98090_DMIC34_A2_LO_SHIFT 0 +#define M98090_DMIC34_A2_LO_WIDTH 8 + +#define M98090_JACK_STATE_NO_HEADSET 0 +#define M98090_JACK_STATE_NO_HEADSET_2 1 +#define M98090_JACK_STATE_HEADPHONE 2 +#define M98090_JACK_STATE_HEADSET 3 + +/* + * M98090_REG_REVISION_ID + */ +#define M98090_REVID_MASK (255<<0) +#define M98090_REVID_SHIFT 0 +#define M98090_REVID_WIDTH 8 +#define M98090_REVID_NUM (1<<M98090_REVID_WIDTH) + +#define M98090_BYTE1(w) ((w >> 8) & 0xff) +#define M98090_BYTE0(w) (w & 0xff) + +/* Silicon revision number */ +#define M98090_REVA 0x40 +#define M98091_REVA 0x50 + +enum max98090_type { + MAX98090, + MAX98091, +}; + +struct max98090_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98090_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + enum max98090_type devtype; + void *control_data; + struct max98090_pdata *pdata; + unsigned int sysclk; + unsigned int bclk; + unsigned int lrclk; + struct max98090_cdata dai[1]; + int irq; + int jack_state; + struct delayed_work jack_work; + struct snd_soc_jack *jack; + unsigned int dai_fmt; + int tdm_slots; + int tdm_width; + u8 lin_state; + unsigned int pa1en; + unsigned int pa2en; + unsigned int extmic_mux; + unsigned int sidetone; +}; + +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 1.7.0.4

1 0

[alsa-devel] crashes in namehint
by Harald Sitter 31 Jan '13

31 Jan '13

aloha, I am working on kde multimedia and we are getting a considerable amount of crash reports from x86 ubuntu and fedora users. https://bugs.kde.org/show_bug.cgi?id=268185 http://notes.kde.org/alsa-bugs these crashes appear to be somewhat limited to x86 with only about 2 reports seeing this happen on x86_64. most of the reports suggest that it happens at startup and in particular for applications that tend to query ALSA devices shortly after startup (in fact, out of main() via a direct call chain). there are 3 different reports of which a sigsev in strcpy and sprintf seem most concerning. both appear in try_config() which was called with a malformed `name` string. now at least the sprintf and the strcpy crashes I can somewhat reliably reproduce on kubuntu 12.10 x86 in a virtualbox vm using the command 'kcmshell4 phonon' with phonon-backend-vlc installed and selected as default. and it appears that those are somehow related to how ubuntu does the pulseaudio integration... in /usr/share/alsa/alsa.conf all of alsa.conf.d/* is included, then there is alsa.conf.d/pulse.conf which contains > hook_func.pulse_load_if_running { > lib "libasound_module_conf_pulse.so" > func "conf_pulse_hook_load_if_running" > } > > @hooks [ > { > func pulse_load_if_running > files [ > "/usr/share/alsa/pulse-alsa.conf" > ] > errors false > } > ] so they load /usr/share/alsa/pulse-alsa.conf after checking PA is running. that file in turn contains the default pulse setup as seen in alsa-plugins [1] now with that lineup I can crash the kcmshell binary about 9/10 times with the strcpy sigsev where name in try_config is 0. if I replace the conditional load in pulse.conf with an explicit one (func load) it crashes about half as often but with the sprintf sigsev in try_config (name now being out-of-bounds rather than 0). if the content of pulse-alsa.conf (i.e. the actual config) is copied into pulse.conf the binary does not crash *at all*. quite the mysterious problem. however, since the crash disappears entirely when removing the second load (i.e. instead of alsa.conf->alsa.conf.d/pulse.conf->pulse-alsa.conf one were to use alsa.conf->alsa.conf.d/pulse.conf), I am lead to believe that this is either a problem caused by the second level of loading in general or a timing problem at large. most likely it is a combination of both where the additional load() calls introduce sufficient overhead to trigger the timing problem. ideas/thoughts/fixes welcome :S HS [1] http://git.alsa-project.org/?p=alsa-plugins.git;a=blob;f=pulse/99-pulseaudi…

3 5

[alsa-devel] [PATCH] ASoC: wm2200: Add ANC source select for outputs
by Chris Rattray 31 Jan '13

31 Jan '13

Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 0181444..a850d88 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1162,6 +1162,14 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, digital_tlv), SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("OUT1L ANC Switch", WM2200_DAC_VOLUME_LIMIT_1L, + WM2200_OUT1L_ANC_SRC_SHIFT, 1, 0), +SOC_SINGLE("OUT1R ANC Switch", WM2200_DAC_VOLUME_LIMIT_1R, + WM2200_OUT1R_ANC_SRC_SHIFT, 1, 0), +SOC_SINGLE("OUT2L ANC Switch", WM2200_DAC_VOLUME_LIMIT_2L, + WM2200_OUT2L_ANC_SRC_SHIFT, 1, 0), +SOC_SINGLE("OUT2R ANC Switch", WM2200_DAC_VOLUME_LIMIT_2R, + WM2200_OUT2R_ANC_SRC_SHIFT, 1, 0), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); -- 1.7.2.5

1 0

[alsa-devel] Previous Post on PandaBoard
by Brent Weatherall 31 Jan '13

31 Jan '13

Hello, My previous post about PandaBoardES audio had one mistake. The code regarding the min/max values was a result of stepping through the 'snd_pcm_hw_params' call and NOT the 'snd_pcm_set_rate_near' call. My apologies for the bad information. Thank you. Regards, Brent

1 0

[alsa-devel] [PATCH 0/3] Make snd_device_name_hint() reentrant
by Jerome Forissier 31 Jan '13

31 Jan '13

The following 3 patches ensure that snd_device_name_hint() does not mdify any global variable. They fix spurious error messages that may occur when the function is called from multiple threads simultaneously. Jerome Forissier (3): Add snd_lib_error_set_local() to install a thread-local error handler. snd_device_name_hint(): do not change the global error handler. snd_device_name_hint(): do not use global snd_config. configure.in | 10 ++++++++ include/error.h | 6 +++++ src/control/namehint.c | 67 ++++++++++++++++++++++++++++---------------------- src/error.c | 24 +++++++++++++++++- 4 files changed, 76 insertions(+), 31 deletions(-) -- 1.8.1.2

1 3

[alsa-devel] [PATCH v3 0/4] ASoC davinci/tlv320aic3x updates, fixes & DT support
by Hebbar Gururaja 31 Jan '13

31 Jan '13

1. Convert tlv320aic3x mic bias to a supply widget and related machine driver 2. Add DT support for Davinci machine platform 3. remove __dev* attributes This patch-set is tested on Davinci platform (DA850 EVM). This series applies on top of tag next-20130128 git tree https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git These patches cannot be tested directly on mainline kernel (DT based). This is because dma is not yet completely ported to Davinci ASoC. I had to add few hacks to get Audio working on Davinci DA850 platform. For reference, I have placed the entire working tree @ Tree : https://github.com/ghebbar/linux.git Branch : dev_next-20130128_davinci_asoc_dt_v3 changes in v3 - drop mcasp pinctrl handler patch from seried as it is now handled centrally. - Convert tlv320aic3x mic bias to a supply widget and related machine driver - remove __dev* attributes Changes in V2 - Remove reference to Linux & software details from DT binding Hebbar Gururaja (3): ASoC: tlv320aic3x: Convert mic bias to a supply widget ASoC: davinci: update machine driver dapm routes ASoC: davinci: remove __dev* attributes Hebbar, Gururaja (1): ASoC: davinci: machine: Add device tree binding .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 6 + include/sound/tlv320aic3x.h | 10 ++ sound/soc/codecs/tlv320aic3x.c | 83 ++++++++- sound/soc/codecs/tlv320aic3x.h | 4 + sound/soc/davinci/davinci-evm.c | 185 +++++++++++++++++--- 6 files changed, 312 insertions(+), 29 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

3 8

[alsa-devel] [PATCH 0/2] Fix thread safety issues
by Jerome Forissier 31 Jan '13

31 Jan '13

This is a set of two patches to fix thread safety issues I found when running a video application over ALSA. The application uses LibVLC to play several videos simultaneously. LibVLC calls snd_pcm_open() and snd_device_name_hint() from multiple threads, without locking--which is allowed according to http://www.alsa-project.org/main/index.php/SMP_Design. Without these patches, I would get random error messages like: ALSA lib confmisc.c:768:(parse_card) cannot find card '$CARD' ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:4663:(snd_config_expand) Evaluate error: No such device Jerome Forissier (2): Make snd_device_name_hint() thread-safe by locking a mutex snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() src/control/namehint.c | 31 +++++++++++++++++++++++++++++++ src/pcm/pcm_direct.c | 15 +++++++++++---- 2 files changed, 42 insertions(+), 4 deletions(-) -- 1.8.1.2

3 8

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Convert mic bias to a supply widget
by Hebbar Gururaja 31 Jan '13

31 Jan '13

Convert MicBias widgets to supply widget. On tlv320aic3x, Mic bias power on/off shares the same register bits with output mic bias voltage. So, when power on mic bias, we need reclaim it to voltage value. Provide a new platform data so that the micbias voltage can be sent according to board requirement. Now since tlc320aic3x codec driver is DT aware, update dt files and functions to handle this new "micbias-vg" platform data. Because of sharing of bits, when enabling the micbias, voltage also needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD macro to create an event to handle this. Signed-off-by: Hebbar Gururaja <gururaja.hebbar(a)ti.com> --- Mark, I have updated the codec driver as you suggested. Can you confirm if this is sufficient?. If yes, then I will send all the patch once again with new revision. :100644 100644 e7b98f4... f47c3f5... M Documentation/devicetree/bindings/sound/tlv320aic3x.txt :100644 100644 ffd9bc7... 9407fd0... M include/sound/tlv320aic3x.h :100644 100644 4989143... fe6dd27... M sound/soc/codecs/tlv320aic3x.c :100644 100644 6db3c41... e521ac3... M sound/soc/codecs/tlv320aic3x.h .../devicetree/bindings/sound/tlv320aic3x.txt | 6 ++ include/sound/tlv320aic3x.h | 10 +++ sound/soc/codecs/tlv320aic3x.c | 85 ++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.h | 4 + 4 files changed, 99 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index e7b98f4..f47c3f5 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -11,6 +11,12 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality +- ai3x-micbias-vg - MicBias Voltage required. + 1 - MICBIAS output is powered to 2.0V, + 2 - MICBIAS output is powered to 2.5V, + 3 - MICBIAS output is connected to AVDD, + If this node is not mentioned or if the value is incorrect, then MicBias + is powered down. Example: diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index ffd9bc7..9407fd0 100644 --- a/include/sound/tlv320aic3x.h +++ b/include/sound/tlv320aic3x.h @@ -46,6 +46,13 @@ enum { AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 }; +enum aic3x_micbias_voltage { + AIC3X_MICBIAS_OFF = 0, + AIC3X_MICBIAS_2_0V = 1, + AIC3X_MICBIAS_2_5V = 2, + AIC3X_MICBIAS_AVDDV = 3, +}; + struct aic3x_setup_data { unsigned int gpio_func[2]; }; @@ -53,6 +60,9 @@ struct aic3x_setup_data { struct aic3x_pdata { int gpio_reset; /* < 0 if not used */ struct aic3x_setup_data *setup; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; #endif diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4989143..fe6dd27 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -85,6 +85,9 @@ struct aic3x_priv { #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 u16 model; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; /* @@ -195,6 +198,37 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, return ret; } +/* + * mic bias power on/off share the same register bits with + * output voltage of mic bias. when power on mic bias, we + * need reclaim it to voltage value. + * 0x0 = Powered off + * 0x1 = MICBIAS output is powered to 2.0V, + * 0x2 = MICBIAS output is powered to 2.5V + * 0x3 = MICBIAS output is connected to AVDD + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + aic3x->micbias_vg << MICBIAS_LEVEL_SHIFT); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, 0); + break; + } + return 0; +} + static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; static const char *aic3x_left_hpcom_mux[] = @@ -596,12 +630,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), /* Mic Bias */ - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", - MICBIAS_CTRL, 6, 3, 1, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", - MICBIAS_CTRL, 6, 3, 2, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", - MICBIAS_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, @@ -1386,6 +1417,24 @@ static int aic3x_probe(struct snd_soc_codec *codec) if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + /* set mic bias voltage */ + switch (aic3x->micbias_vg) { + case AIC3X_MICBIAS_2_0V: + case AIC3X_MICBIAS_2_5V: + case AIC3X_MICBIAS_AVDDV: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + (aic3x->micbias_vg) << MICBIAS_LEVEL_SHIFT); + break; + case AIC3X_MICBIAS_OFF: + /* + * noting to do. target won't enter here. This is just to avoid + * compile time warning "warning: enumeration value + * 'AIC3X_MICBIAS_OFF' not handled in switch" + */ + break; + } + aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); @@ -1461,6 +1510,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; int ret; + u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1474,6 +1524,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (pdata) { aic3x->gpio_reset = pdata->gpio_reset; aic3x->setup = pdata->setup; + aic3x->micbias_vg = pdata->micbias_vg; } else if (np) { ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), GFP_KERNEL); @@ -1493,6 +1544,28 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->setup = ai3x_setup; } + if (!of_property_read_u32(np, "ai3x-micbias-vg", &value)) { + switch (value) { + case 1 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_0V; + break; + case 2 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_5V; + break; + case 3 : + aic3x->micbias_vg = AIC3X_MICBIAS_AVDDV; + break; + default : + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + dev_err(&i2c->dev, "Unsuitable MicBias voltage " + "found in DT\n"); + } + } else { + dev_warn(&i2c->dev, "No MicBias voltage found in DT - " + "using default\n"); + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + } + } else { aic3x->gpio_reset = -1; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6db3c41..e521ac3 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -238,6 +238,10 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* MICBIAS Control Register */ +#define MICBIAS_LEVEL_SHIFT (6) +#define MICBIAS_LEVEL_MASK (3 << 6) + /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) -- 1.7.9.5

4 4

[alsa-devel] [PATCH] Asoc: Samsung: I2S: Add support for runtime S2R
by R. Chandrasekar 31 Jan '13

31 Jan '13

From: "R. Chandrasekar" <rcsekar(a)samsung.com> This patch adds runtime suspend to resume support for I2S. I2S clk is disabled at suspend and enabled at resume. Signed-off-by: R. Chandrasekar <rcsekar(a)samsung.com> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- sound/soc/samsung/i2s.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2fc42f9..d7231e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1071,6 +1071,26 @@ static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) return platform_get_device_id(pdev)->driver_data; } +#ifdef CONFIG_PM_RUNTIME +static int i2s_runtime_suspend(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + + return 0; +} + +static int i2s_runtime_resume(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_prepare_enable(i2s->clk); + + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; @@ -1288,6 +1308,11 @@ static const struct of_device_id exynos_i2s_match[] = { MODULE_DEVICE_TABLE(of, exynos_i2s_match); #endif +static const struct dev_pm_ops samsung_i2s_pm = { + SET_RUNTIME_PM_OPS(i2s_runtime_suspend, + i2s_runtime_resume, NULL) +}; + static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, .remove = samsung_i2s_remove, @@ -1296,6 +1321,7 @@ static struct platform_driver samsung_i2s_driver = { .name = "samsung-i2s", .owner = THIS_MODULE, .of_match_table = of_match_ptr(exynos_i2s_match), + .pm = &samsung_i2s_pm, }, }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: fsl: Fix multiple symbol definitions
by Thierry Reding 31 Jan '13

31 Jan '13

Commit 25b8d31 (ASoC: fsl: fix multiple definition of init_module) fixed a build error due to multiple symbol definitions when building as a module. However, it causes the build to break when the driver is builtin because the imx-pcm.o object is included multiple times. Solve the issue by adding imx-pcm.o to each of the modules that require it, but only add it to the kernel once when one or both drivers are builtin. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/soc/fsl/Makefile | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index ec14579..8446d49 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -30,21 +30,24 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o -snd-soc-imx-pcm-objs := imx-pcm.o -ifneq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),) - snd-soc-imx-pcm-objs += imx-pcm-fiq.o -endif -ifneq ($(CONFIG_SND_SOC_IMX_PCM_DMA),) - snd-soc-imx-pcm-objs += imx-pcm-dma.o -endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o -snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o +snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o +ifeq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),m) +snd-soc-imx-pcm-fiq-y += imx-pcm.o +else +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm.o +endif obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o -snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o +snd-soc-imx-pcm-dma-y := imx-pcm-dma.o +ifeq ($(CONFIG_SND_SOC_IMX_PCM_DMA),m) +snd-soc-imx-pcm-dma-y := imx-pcm.o +else +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm.o +endif # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o -- 1.8.1.2

3 3

[alsa-devel] volume adjust doesn't work after calling many times
by Keyon.Jey 31 Jan '13

31 Jan '13

hi, all, after I use UI interface with pavucontrol to tune output devices volume rapidly many times, the volume adjust seems doesn't work! error occur in dmesg: [ 568.807025] ALSA sound/pci/hda/hda_intel.c:716: azx_get_response timeout, switching to polling mode: last cmd=0x001f000a [ 569.808026] ALSA sound/pci/hda/hda_intel.c:755: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x001f000a realtek ALC888 codec. alsa version 1.0.23 attached the alsa-info output. any comment? BR, Keyon

1 0

Re: [alsa-devel] [PATCH 3/4] ASOC: imx: add machine driver using wm8962 codec
by Mark Brown 31 Jan '13

31 Jan '13

On Thu, Jan 31, 2013 at 03:02:21AM +0000, Zhang Quan-B13634 wrote: > > > + if (!codec_dai->active) > > > + clk_enable(data->codec_clk); > > Should be clk_prepare_enable() but it's not clear to me why you need > > this... Also the clock API does refcounting so there should be no need > > to check for active, you should get matching startups and shutdowns. > > > > This will also fail for analogue bypass paths, set_bias_level() would be a good > > place to cover those. > [Gary-b13634] for power saving consideration. For now remove this feature to > Implement a basic feature I get why you're doing this at all but doing it in startup seems odd - like I say set_bias_level() is the most idiomatic place to implement this usually. There's a few machine drivers doing that you should be able to refer to for examples.

1 0

Re: [alsa-devel] Sound/ARM/i.MX/SSI: fix SDMA starvation
by Shawn Guo 31 Jan '13

31 Jan '13

Thanks for the fix, Juergen. The patch subject could be written in a better form with a proper prefix added. "git log sound/soc/fsl/fsl_ssi.c" should give example. To have the patch get in sooner, you should copy Mark Brown on it. Also MAINTAINERS file gives the following, so you still do not have the right list copied. SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lrg(a)ti.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported F: sound/soc/ F: include/sound/soc* On Wed, Jan 30, 2013 at 02:44:25PM +0100, Juergen Beisert wrote: > Juergen Beisert wrote: > > In cases when capturing is already running and someone enables also > > playback, the SDMA unit of the i.MX SoC does not see an edge on its TX DMA > > request line. This is due to the TX FIFOs are empty, and this request line > > is active all the time, when the CCSR_SSI_SIER_TDMAE bit is set. > > In this case also the TX FIFO underrun interrupt will flood the system > > forever. > > > > To ensure the first edge happens after enabling the TX side of the SSI unit > > and to trigger the SDMA unit successfully enable the CCSR_SSI_SIER_TDMAE on > > demand only. To be sure, this patch does the same for the > > CCSR_SSI_SIER_RDMAE (RX side) bit. > > > > Signed-off-by: Juergen Beisert <jbe(a)pengutronix.de> I can see the patch does fix the problem. A small comment below, otherwise Acked-by: Shawn Guo <shawn.guo(a)linaro.org> > > > > --- > > sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++++++-------- > > 1 file changed, 16 insertions(+), 8 deletions(-) > > > > Index: linux.git/sound/soc/fsl/fsl_ssi.c > > =================================================================== > > --- linux.git.orig/sound/soc/fsl/fsl_ssi.c > > +++ linux.git/sound/soc/fsl/fsl_ssi.c > > @@ -86,10 +86,9 @@ static inline void write_ssi_mask(u32 __ > > #endif > > > > /* SIER bitflag of interrupts to enable */ > > -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ > > - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ > > - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ > > - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ > > +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TIE | \ > > + CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TUE1_EN | \ > > + CCSR_SSI_SIER_RFRC_EN | CCSR_SSI_SIER_RIE | \ It takes me sometime to see the real change here is to remove flags CCSR_SSI_SIER_TDMAE and CCSR_SSI_SIER_RDMAE. Then the changes below should look less dramatic and ease the reviewers a little bit, I think. Shawn --8<--- @@ -86,10 +86,9 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) #endif /* SIER bitflag of interrupts to enable */ -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_RIE | \ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) > > CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) > > > > /** > > @@ -489,6 +488,9 @@ static int fsl_ssi_hw_params(struct snd_ > > * > > * The DMA channel is in external master start and pause mode, which > > * means the SSI completely controls the flow of data. > > + * > > + * To use the i.MX's SDMA unit it is important to create an edge on the > > + * corresponding request line. Otherwise the request will be ignored! > > */ > > static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, > > struct snd_soc_dai *dai) > > @@ -500,20 +502,26 @@ static int fsl_ssi_trigger(struct snd_pc > > switch (cmd) { > > case SNDRV_PCM_TRIGGER_START: > > case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > > - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > > write_ssi_mask(&ssi->scr, 0, > > CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); > > - else > > + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TDMAE); > > + } else { > > write_ssi_mask(&ssi->scr, 0, > > CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); > > + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RDMAE); > > + } > > break; > > > > case SNDRV_PCM_TRIGGER_STOP: > > case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > > - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > > write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); > > - else > > + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TDMAE, 0); > > + } else { > > write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); > > + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RDMAE, 0); > > + } > > break; > > > > default: > > Any comments on this patch? > > jbe > > -- > Pengutronix e.K. | Juergen Beisert | > Linux Solutions for Science and Industry | Phone: +49-5121-206917-5128 | > Peiner Str. 6-8, 31137 Hildesheim, Germany | Fax: +49-5121-206917-5555 | > Amtsgericht Hildesheim, HRA 2686 | http://www.pengutronix.de/ | > > _______________________________________________ > linux-arm-kernel mailing list > linux-arm-kernel(a)lists.infradead.org > http://lists.infradead.org/mailman/listinfo/linux-arm-kernel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH] ASoC: wm2000: Expose ANC gain adjustment
by Mark Brown 30 Jan '13

30 Jan '13

No TLV information since it's not actually a direct gain control. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 4d809ab..f2ac38b 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -666,6 +666,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, } static const struct snd_kcontrol_new wm2000_controls[] = { + SOC_SINGLE("ANC Volume", WM2000_REG_ANC_GAIN_CTRL, 0, 255, 0), SOC_SINGLE_BOOL_EXT("WM2000 ANC Switch", 0, wm2000_anc_mode_get, wm2000_anc_mode_put), @@ -778,6 +779,8 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); + /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix mono
by Benoît Thébaudeau 30 Jan '13

30 Jan '13

The TLV320AIC3x DAI supports only stereo. This patch fixes mono support for this codec by allowing only stereo at codec level. Tested with imx-ssi and TLV320AIC3104. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5708a97..4989143 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1210,13 +1210,13 @@ static struct snd_soc_dai_driver aic3x_dai = { .name = "tlv320aic3x-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, -- 1.7.10.4

2 10

[alsa-devel] Thread safety of snd_pcm_open?
by Jérôme Forissier 29 Jan '13

29 Jan '13

Hello, I'm having problems using ALSA with LibVLC. The project page http://www.alsa-project.org/main/index.php/SMP_Design claims that the snd_*_open() calls are thread safe, yet it seems that when LibVLC calls them from several threads it can lead to errors like: ALSA lib conf.c:4571:(parse_args) Unknown parameter CARD ALSA lib conf.c:4571:(parse_args) Unknown parameter DEV ALSA lib conf.c:4704:(snd_config_expand) Parse arguments error: No such file or directory ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=NVidia,DEV=0 ALSA lib conf.c:4571:(parse_args) Unknown parameter CARD ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=PCH,DEV=0 ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=NVidia,DEV=0 ALSA lib conf.c:4571:(parse_args) Unknown parameter DEV (Please see http://mailman.videolan.org/pipermail/vlc-devel/2013-January/091603.html and also http://mailman.videolan.org/pipermail/vlc-devel/2013-January/091633.html for a reproducer on top of LibVLC). I'm not familiar with the internals of ALSA lib, nor with the internals of LibVLC. The VLC devs claim that they respect the rules regarding ALSA access, but I can't tell for sure... I'm currently running my reproducer with valgrind, but before I dig too deeply, could you please give some advice? I have noticed that pcm_direct.c calls getgrnam() which is not reentrant (but replacing it with getgrnam_r is not enough to fix my problem). Also there does not seem to be any locking around snd_pcm_open[_noupdate]. Are those functions supposed to be reentrant? Thanks for your help, -- Jerome

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix non-snoop page handling
by Takashi Iwai 29 Jan '13

29 Jan '13

For non-snoop mode, we fiddle with the page attributes of CORB/RIRB and the position buffer, but also the ring buffers. The problem is that the current code blindly assumes that the buffer is contiguous. However, the ring buffers may be SG-buffers, thus a wrong vmapped address is passed there, leading to Oops. This patch fixes the handling for SG-buffers. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=800701 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++++++-------------- 1 file changed, 26 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bf0a004..c78286f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -656,29 +656,43 @@ static char *driver_short_names[] = { #define get_azx_dev(substream) (substream->runtime->private_data) #ifdef CONFIG_X86 -static void __mark_pages_wc(struct azx *chip, void *addr, size_t size, bool on) +static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool on) { + int pages; + if (azx_snoop(chip)) return; - if (addr && size) { - int pages = (size + PAGE_SIZE - 1) >> PAGE_SHIFT; + if (!dmab || !dmab->area || !dmab->bytes) + return; + +#ifdef CONFIG_SND_DMA_SGBUF + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) { + struct snd_sg_buf *sgbuf = dmab->private_data; if (on) - set_memory_wc((unsigned long)addr, pages); + set_pages_array_wc(sgbuf->page_table, sgbuf->pages); else - set_memory_wb((unsigned long)addr, pages); + set_pages_array_wb(sgbuf->page_table, sgbuf->pages); + return; } +#endif + + pages = (dmab->bytes + PAGE_SIZE - 1) >> PAGE_SHIFT; + if (on) + set_memory_wc((unsigned long)dmab->area, pages); + else + set_memory_wb((unsigned long)dmab->area, pages); } static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, bool on) { - __mark_pages_wc(chip, buf->area, buf->bytes, on); + __mark_pages_wc(chip, buf, on); } static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, - struct snd_pcm_runtime *runtime, bool on) + struct snd_pcm_substream *substream, bool on) { if (azx_dev->wc_marked != on) { - __mark_pages_wc(chip, runtime->dma_area, runtime->dma_bytes, on); + __mark_pages_wc(chip, snd_pcm_get_dma_buf(substream), on); azx_dev->wc_marked = on; } } @@ -689,7 +703,7 @@ static inline void mark_pages_wc(struct azx *chip, struct snd_dma_buffer *buf, { } static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, - struct snd_pcm_runtime *runtime, bool on) + struct snd_pcm_substream *substream, bool on) { } #endif @@ -1968,11 +1982,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; - struct snd_pcm_runtime *runtime = substream->runtime; struct azx_dev *azx_dev = get_azx_dev(substream); int ret; - mark_runtime_wc(chip, azx_dev, runtime, false); + mark_runtime_wc(chip, azx_dev, substream, false); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; azx_dev->format_val = 0; @@ -1980,7 +1993,7 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, params_buffer_bytes(hw_params)); if (ret < 0) return ret; - mark_runtime_wc(chip, azx_dev, runtime, true); + mark_runtime_wc(chip, azx_dev, substream, true); return ret; } @@ -1989,7 +2002,6 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx_dev *azx_dev = get_azx_dev(substream); struct azx *chip = apcm->chip; - struct snd_pcm_runtime *runtime = substream->runtime; struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; /* reset BDL address */ @@ -2002,7 +2014,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) snd_hda_codec_cleanup(apcm->codec, hinfo, substream); - mark_runtime_wc(chip, azx_dev, runtime, false); + mark_runtime_wc(chip, azx_dev, substream, false); return snd_pcm_lib_free_pages(substream); } -- 1.8.1.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable LPIB delay count for Poulsbo / Oaktrail
by Takashi Iwai 29 Jan '13

29 Jan '13

Currently we use LPIB forcibly for both playback and capture for Poulsbo and Oaktrail devices, and this seems rather problematic. The recent fix for LPIB delay count seems working well with these devices, so let's enable it instead. Reported-by: Martin Weishart <martin.weishart(a)telosalliance.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0b6aeba..bf0a004 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3613,13 +3613,12 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* 5 Series/3400 */ { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH }, - /* SCH */ + /* Poulsbo */ { PCI_DEVICE(0x8086, 0x811b), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* Oaktrail */ { PCI_DEVICE(0x8086, 0x080a), - .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Oaktrail */ + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- 1.8.1.1

1 0

[alsa-devel] [PATCH 0/6] Misc ALSA changes, documentation, style
by Antonio Ospite 29 Jan '13

29 Jan '13

From: Antonio Ospite <ao2(a)amarulasolutions.com> Hi, here are some fixes for things I noticed while studying ALSA. I'd like some feedback especially on patch 1. Patches 2 and 3 are documentation fixes. Patches 4, 5 and 6, are basically style fixes. Feel free to discard patch 6 if you find it too invasive, it's just cosmetics but some maintainers don't like big style changes even if they are in the "right" direction. I don't know alsa devs' position about that. I am sending the whole series to alsa-devel and CCing topic specific mailing lists or developers on a per-patch basis. Thanks, Antonio Antonio Ospite (6): Force a cast to silence a warning from "sparse" Documentation/DocBook/writing-an-alsa-driver.tmpl: fix some thinkos Documentation/DocBook/writing-an-alsa-driver.tmpl: fix some typos sound/usb/caiaq/device.c: fix use of MODULE_SUPPORTED_DEVICES() sound/usb/card.c: cosmetics, remove a leading space sound/core/pcm_native.c: cosmetics, fix use of spaces Documentation/DocBook/writing-an-alsa-driver.tmpl | 11 +- include/sound/memalloc.h | 2 +- sound/core/pcm_native.c | 120 ++++++++++----------- sound/usb/caiaq/device.c | 8 +- sound/usb/card.c | 2 +- 5 files changed, 71 insertions(+), 72 deletions(-) -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

3 11

Re: [alsa-devel] [PATCH 1/4] DMA: PL330: Add xlate function
by Arnd Bergmann 29 Jan '13

29 Jan '13

On Friday 18 January 2013, Padmavathi Venna wrote: > +struct dma_chan *of_dma_pl330_xlate(struct of_phandle_args *dma_spec, > + struct of_dma *ofdma) > +{ > + int count = dma_spec->args_count; > + struct of_dma_filter_info *info = ofdma->of_dma_data; > + > + if (!info || !info->filter_fn) > + return NULL; > + > + if (count != 1) > + return NULL; > + > + return dma_request_channel(info->dma_cap, info->filter_fn, dma_spec); > +} > +EXPORT_SYMBOL_GPL(of_dma_pl330_xlate); It seems a little sad that we still have to use dma_request_channel() to implement this, when that function will go off searching all channels and pass them tino the filter, which then has to look for the device node and match it with each channel. We already know the controller and should just be able to get a channel for it, although I don't exactly know how that is done. Further, your function is almost identical to the of_dma_simple_xlate function. Can't you use that one instead? Arnd

2 2

Re: [alsa-devel] [PATCH 4/4] ASOC: wm8962: add some operation in wm8962_runtime_resume()
by Mark Brown 29 Jan '13

29 Jan '13

On Tue, Jan 29, 2013 at 03:50:10PM +0800, Gary Zhang wrote: > 1. in this function, after wm8962_reset() is called, WM8962_SYSCLK_ENA bit > is reset to 1. make sure it is off so we can safely write to registers if > the device is declocked. > 2. add bias level management Split separate changes into separate patches; obviously there's bias level management in the driver since otherwise it'd be totally non-functional. > @@ -3721,6 +3721,24 @@ static int wm8962_runtime_resume(struct device *dev) > > regcache_sync(wm8962->regmap); > > + regmap_update_bits(wm8962->regmap, WM8962_CLOCKING2, > + WM8962_SYSCLK_ENA, 0); > + This seems like a bodge to work around some other problem in your system. The device should never be runtime suspended with an active clock, and this seems like it isn't going to work as most of the register cache restore will be done with SYSCLK enabled. We need to figure out what went wrong to allow the device to enter runtime suspend with SYSCLK on. > + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, > + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, > + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); > + > + /* Bias enable at 2*50k for ramp */ > + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, > + WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, > + WM8962_BIAS_ENA | 0x180); > + > + msleep(5); > + > + /* VMID back to 2x250k for standby */ > + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, > + WM8962_VMID_SEL_MASK, 0x100); We already have code to ramp VMID when exiting _OFF...

1 0

Re: [alsa-devel] [PATCH V6 07/10] ASoC: SMDK: WM8994: Add device tree support for machine file
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:06PM +0530, Padmavathi Venna wrote: > +Samsung SMDK audio complex This is just for SMDKs with WM8994. I'll apply but please send a followup patch to clarify this - it'll be a different binding for others like the older boards using WM8580 and WM9713. > + samsung,i2s-controller = <&i2s0>; > + samsung,audio-codec = <&wm8994>; I was thinking this should really point at the DAI but since it's specific to the SMDKs I don't think it particularly matters - they all use AIF1 anyway.

2 1

Re: [alsa-devel] [PATCH V6 04/10] spi: s3c64xx: Modify SPI driver to use generic DMA DT support
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:03PM +0530, Padmavathi Venna wrote: > This patch modifies the SPI driver to use generic dma dt bindings > support. This passes all the required arguments to dma dev request > functon which in turn calls the dma_request_slave_channel or dma__ > request_channel based on DT or non-DT respectively. This loooks OK and I'm actually applying SPI patches so I could apply it but I'm not sure I see the relevance of this patch to the rest of the series (which is about ASoC). Is there some reason why it's included in this patch series or can it be applied to the SPI tree? The SMDKs use I2C for their CODEC control rather than SPI. Also is there a binding document update or device tree file update to go with this? > + if (!sdd->pdev->dev.of_node) { > + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); > + if (!res) { > + dev_err(&pdev->dev, "Unable to get SPI tx dma " > + "resource\n"); I appreciate that this is cut'n'paste from the code you're refactoring but please don't split error messages over lines, it makes it hard to grep for them in the kernel source.

2 1

Re: [alsa-devel] [PATCH V6 03/10] ARM: SAMSUNG: Make dma request compatible to generic dma bindings.
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:02PM +0530, Padmavathi Venna wrote: > This patch make the dma dev request operation compatible for both > DT and non-DT cases. It takes the all the arguments required for > dma_request_slave_channel and dma_request_channel. If the driver > is initiated via DT or non-DT the corresponding call will be made. OK, so this is where the SPI change came from - the API has been changed incompatibly and SPI needs updating. For bisection this API update ought to be done in a single commit, otherwise there will be steps in the bisection where nothing will work. Anyway, I'll go ahead and apply this, the SPI and the ASoC ones but this is something to bear in mind for the future.

2 1

Re: [alsa-devel] [PATCH 3/4] ASOC: imx: add machine driver using wm8962 codec
by Mark Brown 29 Jan '13

29 Jan '13

On Tue, Jan 29, 2013 at 03:49:50PM +0800, Gary Zhang wrote: > +static struct imx_priv card_priv; Global data bad... > +static int imx_wm8962_startup(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_dai *codec_dai = rtd->codec_dai; > + struct imx_priv *priv = &card_priv; > + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); > + > + if (!codec_dai->active) > + clk_enable(data->codec_clk); Should be clk_prepare_enable() but it's not clear to me why you need this... Also the clock API does refcounting so there should be no need to check for active, you should get matching startups and shutdowns. This will also fail for analogue bypass paths, set_bias_level() would be a good place to cover those. > + int_port--; > + ext_port--; > + ret = imx_audmux_v2_configure_port(int_port, > + IMX_AUDMUX_V2_PTCR_SYN | > + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | > + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | > + IMX_AUDMUX_V2_PTCR_TFSDIR | > + IMX_AUDMUX_V2_PTCR_TCLKDIR, > + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); > + if (ret) { > + dev_err(&pdev->dev, "audmux internal port setup failed\n"); > + return ret; > + } > + imx_audmux_v2_configure_port(ext_port, > + IMX_AUDMUX_V2_PTCR_SYN, > + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); > + if (ret) { > + dev_err(&pdev->dev, "audmux external port setup failed\n"); > + return ret; > + } This stuff is pretty common for i.MX drivers - can we factor it out into a helper library? > + ssi_pdev = of_find_device_by_node(ssi_np); > + if (!ssi_pdev) { > + dev_err(&pdev->dev, "failed to find SSI platform device\n"); > + ret = -EINVAL; > + goto fail; > + } Please remember to include binding documentation for all new device tree bindings.

1 0

Re: [alsa-devel] [PATCH 3/4] ARM: dts: Add #dma-cells for generic dma binding support
by Arnd Bergmann 29 Jan '13

29 Jan '13

On Friday 18 January 2013, Padmavathi Venna wrote: > diff --git a/Documentation/devicetree/bindings/dma/arm-pl330.txt b/Documentation/devicetree/bindings/dma/arm-pl330.txt > index 36e27d5..457a233 100644 > --- a/Documentation/devicetree/bindings/dma/arm-pl330.txt > +++ b/Documentation/devicetree/bindings/dma/arm-pl330.txt > @@ -8,6 +8,8 @@ Required properties: > - reg: physical base address of the controller and length of memory mapped > region. > - interrupts: interrupt number to the cpu. > + - #dma-cells: must be at least 1. used to represent the number of integer > + cells in the dmas property of client device. The wording 'at least' seems wrong here: that is what we use in the generic DMA binding, but in the part that is specific to one driver, I would expect to see - #dma-cells: must be <1> Since that is what this particular driver requires. > Client drivers (device nodes requiring dma transfers from dev-to-mem or > @@ -27,7 +30,7 @@ as shown below. > [property name] = <[phandle of the dma controller] [dma request id]>; > > where 'dma request id' is the dma request number which is connected > - to the client controller. The 'property name' is recommended to be > - of the form <name>-dma-channel. > + to the client controller. The 'property name' is 'dmas' as recommended > + by the generic dma device tree binding helpers. s/recommended/required/ Also, the "dma-names" property is required as well. Arnd

2 1

Re: [alsa-devel] [PATCH V6 10/10] dmaengine: Fix compilation error in non-DT case
by Vinod Koul 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:09PM +0530, Padmavathi Venna wrote: > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > --- > include/linux/dmaengine.h | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h > index 8cd0e25..c88f302 100644 > --- a/include/linux/dmaengine.h > +++ b/include/linux/dmaengine.h > @@ -992,7 +992,7 @@ static inline struct dma_chan *__dma_request_channel(dma_cap_mask_t *mask, > static inline struct dma_chan *dma_request_slave_channel(struct device *dev, > char *name) > { > - return NULL > + return NULL; what tree was this generated against? this was fixed by 678bd8 -- ~Vinod

2 1

Re: [alsa-devel] [PATCH V6 09/10] ASoC: Samsung: Update Kconfig for I2S, SPDIF and PCM audio
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:08PM +0530, Padmavathi Venna wrote: > Update Kconfig file to enable I2S,PCM audio for wm8994 and spdif on all > samsung platforms. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH V6 08/10] ARM: dts: Modify SPI nodes according generic DMA DT bindings
by Mark Brown 29 Jan '13

29 Jan '13

On Fri, Jan 18, 2013 at 05:17:07PM +0530, Padmavathi Venna wrote: > This patch removes custom way of adding spi dma channels and > adds according to new generic DMA DT bindings. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/4] DMA: PL330: Register the DMA controller with the generic DMA helpers
by Arnd Bergmann 28 Jan '13

28 Jan '13

On Monday 28 January 2013, Jon Hunter wrote: > On 01/28/2013 08:51 AM, Arnd Bergmann wrote: > > On Friday 18 January 2013, Padmavathi Venna wrote: > >> + pfi = devm_kzalloc(&adev->dev, sizeof(*pfi), GFP_KERNEL); > >> + if (!pfi) { > >> + dev_err(&adev->dev, "unable to allocate mem\n"); > >> + return -ENOMEM; > >> + } > >> + > >> + pfi->dma_cap = pd->cap_mask; > >> + pfi->filter_fn = pl330_filter; > >> + > >> + ret = of_dma_controller_register(adev->dev.of_node, > >> + of_dma_pl330_xlate, pfi); > > > > Why do you pass a 'struct of_dma_filter_info' here? I would > > expect that you pass the pdmac object as the third argument. > > I believe it is because that is the data that the xlate function is > using. Are you suggesting the data should be stored in the pdmac object > and extracted from there? That could be done too given that this > controller has its own xlate. It just seems weird that we are passing a constant cap_mask in a data structure, and that we need our own filter function still, but don't pass the pointer to the data structure that we actually need in the filter function (the dma_device *). Arnd

1 0

Re: [alsa-devel] [PATCH 0/4] Add generic DMA DT binding support
by Vinod Koul 28 Jan '13

28 Jan '13

On Fri, Jan 18, 2013 at 05:03:40PM +0530, Padmavathi Venna wrote: > This patch set adds support for generic dma device tree bindings for > Samsung platforms and is dependent on the following patches from > Vinod Koul next branch > 1)of: Add generic device tree DMA helpers > 2)dmaengine: add helper function to request a slave DMA channel Changes look fairly decent. I need somone with better knowldge of DT to akc this before this is applied. Arnd...? > > This patch set is made based Kukjin Kim for-next branch > > Padmavathi Venna (4): > DMA: PL330: Add xlate function > DMA: PL330: Register the DMA controller with the generic DMA helpers > ARM: dts: Add #dma-cells for generic dma binding support > DMA: PL330: Modify pl330 filter based on new generic dma dt bindings. > > .../devicetree/bindings/dma/arm-pl330.txt | 9 +++- > arch/arm/boot/dts/exynos5250.dtsi | 4 ++ > drivers/dma/pl330.c | 48 ++++++++++++++++---- > 3 files changed, 49 insertions(+), 12 deletions(-) > > -- > 1.7.4.4 >

2 1

Re: [alsa-devel] [PATCH 2/4] DMA: PL330: Register the DMA controller with the generic DMA helpers
by Arnd Bergmann 28 Jan '13

28 Jan '13

On Friday 18 January 2013, Padmavathi Venna wrote: > + pfi = devm_kzalloc(&adev->dev, sizeof(*pfi), GFP_KERNEL); > + if (!pfi) { > + dev_err(&adev->dev, "unable to allocate mem\n"); > + return -ENOMEM; > + } > + > + pfi->dma_cap = pd->cap_mask; > + pfi->filter_fn = pl330_filter; > + > + ret = of_dma_controller_register(adev->dev.of_node, > + of_dma_pl330_xlate, pfi); Why do you pass a 'struct of_dma_filter_info' here? I would expect that you pass the pdmac object as the third argument. Arnd

1 0

[alsa-devel] Some information about UsbPre2 UAC2 support
by shal＠free.fr 28 Jan '13

28 Jan '13

Hi all, The Sound Devices USBPre2 is an UAC2 device but recognized as a UAC1 device by Linux. I have spend some time in order to found a solution. I have successfully setup my sound card as a UAC2 mode and I can play/record a 24/192khz without problem. But my solution is very ugly and cannot be considered as a real solution. Perhaps alsa developers have an idea to integrated UAC2 support for the USBPRe2 in the standard stack.... There is two distinct problems: - The sound card recognizes Linux as a Windows OS and send a incorrect device descriptor. - The USB stack choose the bad configuration descriptor. More details on first problem: For some compatibility raison, Sound device developer want only send a UAC1 configuration when the Operating system is Windows. The used method is based on the length of the setup packet during the device get_descriptor function. If the length is 64, the OS is considered as Windows. Linux have choosen also this value. Extract of kernel source code: " NOTE: If USE_NEW_SCHEME() is true we will start by issuing a 64-byte GET_DESCRIPTOR request. This is what Windows does, so it may help with some non-standards-compliant devices." So I have patched my kernel. In driver/usb/core/hub.c: #define GET_DESCRIPTOR_BUFSIZE 65 (instead 64) This solve the first problem. More details on the second problem: Now, the USB device descriptor has two configuration descriptors :the first is a UAC1 and the second is a UAC2. My "lsusb -v -d 0926:" can be view at: http://pastebin.com/nFTN2E04 As UAC1 is a valid configuration descriptor, the usb stack choose this configuration. I have patched the usb_choose_configuration() function in file driver/usb/core/generic.c for overwrite the value with the good one. That all, no other problem was detected. The Linux UAC2 stack seems to be OK with this sound card. I joined (ugly) patches , built on linux 3.7.1. If you have an idea about how have a standard support of UAC2 for the Sound Devices USBPRe2, I (and a lot of people) will be happy. Thanks Olivier PS: A thread exists on LAU : http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-January/089199.…

2 5

[alsa-devel] [PATCH] ALSA: hda - Fix powermap for external mics on IDT codecs
by David Henningsson 28 Jan '13

28 Jan '13

This patch fixes a regression of the external mic not working on HP Probook 4520s. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 8 ++++++++ 1 file changed, 8 insertions(+) This is why my external mic stopped working; I now had some time to track it down. Alsa-info at http://paste.ubuntu.com/1579993/ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1c49861..951fc0d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -502,6 +502,13 @@ static void stac_line_automute(struct hda_codec *codec, jack_update_power(codec, jack); } +static void stac_mic_autoswitch(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + snd_hda_gen_mic_autoswitch(codec, jack); + jack_update_power(codec, jack); +} + static void stac_vref_event(struct hda_codec *codec, struct hda_jack_tbl *event) { unsigned int data; @@ -3509,6 +3516,7 @@ static int stac_parse_auto_config(struct hda_codec *codec) spec->gen.automute_hook = stac_update_outputs; spec->gen.hp_automute_hook = stac_hp_automute; spec->gen.line_automute_hook = stac_line_automute; + spec->gen.mic_autoswitch_hook = stac_mic_autoswitch; err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ucm: Set uc_mgr->ctl to NULL after closing it.
by Tanu Kaskinen 28 Jan '13

28 Jan '13

Fixes a double-free bug. Signed-off-by: Tanu Kaskinen <tanuk(a)iki.fi> --- src/ucm/main.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/ucm/main.c b/src/ucm/main.c index 0d7d841..d8b8ada 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -145,6 +145,7 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, free(uc_mgr->ctl_dev); uc_mgr->ctl_dev = NULL; snd_ctl_close(uc_mgr->ctl); + uc_mgr->ctl = NULL; } err = snd_ctl_open(ctl, ctl_dev, 0); -- 1.7.10.4

3 2

Re: [alsa-devel] 3.7 HDMI channel map regression
by Takashi Iwai 28 Jan '13

28 Jan '13

At Sun, 27 Jan 2013 19:18:27 -0600, Shawn Bohrer wrote: > > Hi Takashi, > > I recently updated my HTPC from 3.6.11 to 3.7.2 and this caused my RL > and FC channels to swap, and my RR and LFE channels to swap for PCM > audio. Doing a git bisect identified > d45e6889ee69456a4d5b1bbb32252f460cd48fa9 "ALSA: hda - Provide the > proper channel mapping for generic HDMI driver" as the commit that > caused my channels to swap. The commit doesn't revert cleanly on > 3.7.4, and I haven't really looked to see what the correct fix might > be. > > Some info that may be relevant, the sound card is a: > > 00:1b.0 Audio device: Intel Corporation 7 Series/C210 Series Chipset > Family High Definition Audio Controller (rev 04) > > The machine is running Fedora 18 and audio goes over HDMI to a 5.1 > receiver. I'm not really sure what other info you might need, but > let me know if you need something else or have any patches you would > like me to test. OK, it's the first time to get a bug report about this. Could you tell me how did you test it (i.e. which application, which sound backend)? Can you confirm that it's reproduced via speaker-test program in alsa-utils package? For further debugging, please give the following: - alsa-info.sh output while playing 5.1 sound - /proc/asound/card*/eld* contents while playing 5.1 sound Also, try to update alsa-lib and alsa-utils from alsa-project.org git tree, and build alsa-lib/test/chmap program. % cd alsa-lib/test % make chmap and run like ./chmap -Dhdmi:1 query (-Dhdmi:1 might be different depending on the system setup) thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - fix inverted internal mic on Acer AOA150/ZG5
by David Henningsson 28 Jan '13

28 Jan '13

This patch enables internal mic input on the machine. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1107477 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dc75607..de512c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2363,6 +2363,7 @@ static const struct hda_model_fixup alc268_fixup_models[] = { }; static const struct snd_pci_quirk alc268_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x015b, "Acer AOA 150 (ZG5)", ALC268_FIXUP_INV_DMIC), /* below is codec SSID since multiple Toshiba laptops have the * same PCI SSID 1179:ff00 */ -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH 3/3] ALSA: intel8x0: Check return value before assigning
by Takashi Iwai 28 Jan '13

28 Jan '13

At Sun, 27 Jan 2013 17:40:02 +0100, Emil Goode wrote: > > The first patch in this series changes the return type of function > kvm_para_available to bool. In the function snd_intel8x0_inside_vm > we now need to check it's return value before assigning a value to > the result variable. > > Signed-off-by: Emil Goode <emilgoode(a)gmail.com> > --- > sound/pci/intel8x0.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c > index 3b9be75..fa9337e 100644 > --- a/sound/pci/intel8x0.c > +++ b/sound/pci/intel8x0.c > @@ -2982,7 +2982,7 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) > } > > /* detect KVM and Parallels virtual environments */ > - result = kvm_para_available(); > + result = kvm_para_available() ? 1 : 0; This is superfluous. The conversion from bool to int works implicitly like that. Takashi

1 0

[alsa-devel] [PATCH 1/2] Revert "ASoC: fsl: fix multiple definition of init_module"
by Shawn Guo 28 Jan '13

28 Jan '13

This reverts commit 25b8d31488a3fb3611651991969526b2ea475764. While the commit fixes multiple init_module definition error with module build, it breaks build when both imx-pcm-fiq and imx-pcm-dma are built in as below. LD sound/soc/fsl/snd-soc-fsl-ssi.o LD sound/soc/fsl/snd-soc-fsl-utils.o LD sound/soc/fsl/snd-soc-imx-ssi.o LD sound/soc/fsl/snd-soc-imx-audmux.o LD sound/soc/fsl/snd-soc-imx-pcm-fiq.o LD sound/soc/fsl/snd-soc-imx-pcm-dma.o LD sound/soc/fsl/snd-soc-eukrea-tlv320.o LD sound/soc/fsl/snd-soc-imx-sgtl5000.o LD sound/soc/fsl/snd-soc-imx-mc13783.o LD sound/soc/fsl/built-in.o sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_free': imx-pcm.c:(.text+0x464): multiple definition of `imx_pcm_free' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x1a8): first defined here sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `snd_imx_pcm_mmap': imx-pcm.c:(.text+0x35c): multiple definition of `snd_imx_pcm_mmap' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0xa0): first defined here sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_new': imx-pcm.c:(.text+0x3dc): multiple definition of `imx_pcm_new' sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x120): first defined here make[4]: *** [sound/soc/fsl/built-in.o] Error 1 Let's revert the commit and find a proper fix for multiple init_module definition error later. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/fsl/Kconfig | 9 +++++++-- sound/soc/fsl/Makefile | 5 +---- sound/soc/fsl/imx-pcm.c | 3 +++ 3 files changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a210c8d..3b98159 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,13 +108,18 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM_FIQ +config SND_SOC_IMX_PCM tristate + +config SND_SOC_IMX_PCM_FIQ + bool select FIQ + select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - tristate + bool select SND_SOC_DMAENGINE_PCM + select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index ec14579..afd3479 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -41,10 +41,7 @@ endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o -snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o -obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o -snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index 0c9f188..d5cd9ef 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -31,6 +31,7 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); return ret; } +EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -79,6 +80,7 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } +EXPORT_SYMBOL_GPL(imx_pcm_new); void imx_pcm_free(struct snd_pcm *pcm) { @@ -100,6 +102,7 @@ void imx_pcm_free(struct snd_pcm *pcm) buf->area = NULL; } } +EXPORT_SYMBOL_GPL(imx_pcm_free); MODULE_DESCRIPTION("Freescale i.MX PCM driver"); MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); -- 1.7.9.5

2 2

[alsa-devel] [PATCH - TINYCOMPRESS 1/4] compress: Extend limit on error messages
by Charles Keepax 28 Jan '13

28 Jan '13

32 characters is a very short size for the error message, extend this to 128 characters to prevent clipping on the majority of messages. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> diff --git a/compress.c b/compress.c index faef982..05e54ae 100644 --- a/compress.c +++ b/compress.c @@ -75,7 +75,7 @@ #include "sound/compress_offload.h" #include "tinycompress/tinycompress.h" -#define COMPR_ERR_MAX 32 +#define COMPR_ERR_MAX 128 struct compress { int fd; -- 1.7.2.5

2 4

Re: [alsa-devel] [PATCH 3/3] ALSA: intel8x0: Check return value before assigning
by H. Peter Anvin 27 Jan '13

27 Jan '13

Wtf? This patch seems semibogus at best and the description is even more ridiculous... Emil Goode <emilgoode(a)gmail.com> wrote: >The first patch in this series changes the return type of function >kvm_para_available to bool. In the function snd_intel8x0_inside_vm >we now need to check it's return value before assigning a value to >the result variable. > >Signed-off-by: Emil Goode <emilgoode(a)gmail.com> >--- > sound/pci/intel8x0.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > >diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c >index 3b9be75..fa9337e 100644 >--- a/sound/pci/intel8x0.c >+++ b/sound/pci/intel8x0.c >@@ -2982,7 +2982,7 @@ static int snd_intel8x0_inside_vm(struct pci_dev >*pci) > } > > /* detect KVM and Parallels virtual environments */ >- result = kvm_para_available(); >+ result = kvm_para_available() ? 1 : 0; > #ifdef X86_FEATURE_HYPERVISOR > result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); > #endif -- Sent from my mobile phone. Please excuse brevity and lack of formatting.

1 0

[alsa-devel] ASoC updates for v3.8
by Mark Brown 27 Jan '13

27 Jan '13

The following changes since commit 7d1f9aeff1ee4a20b1aeb377dd0f579fe9647619: Linux 3.8-rc4 (2013-01-17 19:25:45 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.8-rc4 for you to fetch changes up to a8c136d2eba7b0b0c872e59eb404b1fe36504112: Merge remote-tracking branch 'asoc/fix/wm2200' into tmp (2013-01-22 16:26:33 +0800) ---------------------------------------------------------------- ASoC: Updates for v3.8-rc4 The usual set of driver updates, nothing too thrilling in here - one core change for the regulator bypass mode which was just not doing the right thing at all and a bunch of driver specifics. ---------------------------------------------------------------- Charles Keepax (2): ASoC: arizona: Disable free-running mode on FLL1 ASoC: wm_adsp: Release firmware on error Chris Rattray (1): ASoC: wm2200: correct mixer values and text Liam Girdwood (1): ASoC: MAINTAINERS: Update email address. Mark Brown (10): ASoC: dapm: Fix sense of regulator bypass mode ASoC: wm5102: Correct AEC loopback mask ASoC: wm5110: Correct AEC loopback mask ASoC: arizona: Use actual rather than desired BCLK when calculating LRCLK ASoC: wm_adsp: Use GFP_DMA for things that may be DMAed Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/fsl' into tmp Merge remote-tracking branch 'asoc/fix/lrg' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Shawn Guo (1): ASoC: fsl: fix multiple definition of init_module MAINTAINERS | 2 +- sound/soc/codecs/arizona.c | 5 ++++- sound/soc/codecs/wm2200.c | 3 --- sound/soc/codecs/wm5102.c | 3 +-- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm_adsp.c | 6 +++--- sound/soc/fsl/Kconfig | 9 ++------- sound/soc/fsl/Makefile | 5 ++++- sound/soc/fsl/imx-pcm.c | 3 --- sound/soc/soc-dapm.c | 12 ++++++++++-- 10 files changed, 26 insertions(+), 25 deletions(-)

2 1

[alsa-devel] [git pull] usb-audio: RME Babyface/Fireface fix
by Clemens Ladisch 27 Jan '13

27 Jan '13

The following change since commit 29594404d7fe73cd80eaa4ee8c43dcc53970c60e: Linux 3.7 (2012-12-10 19:30:57 -0800) is available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git usb-audio-fix for you to fetch changes up to 5fcbad428a81efa318f15bfcfbcd8f0a98ec0b11: ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices (2013-01-26 17:49:35 +0100) ---------------------------------------------------------------- Clemens Ladisch (1): ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices sound/usb/mixer.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices Commit 23caaf19b11e (ALSA: usb-mixer: Add support for Audio Class v2.0) forgot to adjust the length check for UAC 2.0 feature unit descriptors. This would make the code abort on encountering a feature unit without per-channel controls, and thus prevented the driver to work with any device having such a unit, such as the RME Babyface or Fireface UCX. Reported-by: Florian Hanisch <fhanisch(a)uni-potsdam.de> Tested-by: Matthew Robbetts <wingfeathera(a)gmail.com> Tested-by: Michael Beer <beerml(a)sigma6audio.de> Cc: Daniel Mack <daniel(a)caiaq.de> Cc: 2.6.35+ <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 298070e..41e8bfb 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1259,16 +1259,23 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void } channels = (hdr->bLength - 7) / csize - 1; bmaControls = hdr->bmaControls; + if (hdr->bLength < 7 + csize) { + snd_printk(KERN_ERR "usbaudio: unit %u: " + "invalid UAC_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } } else { struct uac2_feature_unit_descriptor *ftr = _ftr; csize = 4; channels = (hdr->bLength - 6) / 4 - 1; bmaControls = ftr->bmaControls; - } - - if (hdr->bLength < 7 || !csize || hdr->bLength < 7 + csize) { - snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); - return -EINVAL; + if (hdr->bLength < 6 + csize) { + snd_printk(KERN_ERR "usbaudio: unit %u: " + "invalid UAC_FEATURE_UNIT descriptor\n", + unitid); + return -EINVAL; + } } /* parse the source unit */

2 1

Re: [alsa-devel] [PATCH v2] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Mark Brown 27 Jan '13

27 Jan '13

On Fri, Jan 18, 2013 at 10:17:00AM +0100, Michal Bachraty wrote: > u32 rotate = (32 - word_length) / 4; > This implementation is wrong, but it works only for 16, or 32 bit audio data. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH V6 10/10] dmaengine: Fix compilation error in non-DT case
by Mark Brown 27 Jan '13

27 Jan '13

On Fri, Jan 18, 2013 at 05:17:09PM +0530, Padmavathi Venna wrote: > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> This should be sent separately to Vinod for merge into v3.8, it's not obviously related to the rest of the series and is a bug fix.

1 0

[alsa-devel] upsampling
by Ran Shalit 26 Jan '13

26 Jan '13

Hello, I would like to ask if I can use alsa-lib to resample (downsample or sample) input audio and playback audio. I searched for appropriate API but found none. Regards, Ran

2 1

[alsa-devel] [patch] ASoC: dwc: fix support for more than two channels
by Dan Carpenter 26 Jan '13

26 Jan '13

There were missing break statements so everything used TWO_CHANNEL_SUPPORT. Also I added a return statement to silence a GCC warning: sound/soc/dwc/designware_i2s.c: In function ‘dw_i2s_hw_params’: sound/soc/dwc/designware_i2s.c:236:32: warning: ‘ch_reg’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Static analysis. Untested. diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 1aa5130..deb30d5 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -210,15 +210,19 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, switch (config->chan_nr) { case EIGHT_CHANNEL_SUPPORT: ch_reg = 3; + break; case SIX_CHANNEL_SUPPORT: ch_reg = 2; + break; case FOUR_CHANNEL_SUPPORT: ch_reg = 1; + break; case TWO_CHANNEL_SUPPORT: ch_reg = 0; break; default: dev_err(dev->dev, "channel not supported\n"); + return -EINVAL; } i2s_disable_channels(dev, substream->stream);

3 2

Re: [alsa-devel] [PATCH] Asoc: Samsung: Add support for I2S runtime S2R
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 06:05:30PM +0530, R. Chandrasekar wrote: > From: "R. Chandrasekar" <rcsekar(a)samsung.com> > > This patch adds runtime suspend to resume support for I2S. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > Signed-off-by: R. Chandrasekar <rcsekar(a)samsung.com> This is good but doesn't apply against topic/samsung, please check and resend. Note that the signoffs should be in author order - the author should be first, then people forwarding on the patch. Other people should have other tags like Acked-by or Reviewed-by.

1 0

Re: [alsa-devel] [PATCH] Asoc: Samsung: Add I2S S/W RST in startup function
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 06:05:31PM +0530, R. Chandrasekar wrote: > From: Padmavathi Venna <padma.v(a)samsung.com> > > I2S module need to be reset after S2R. Keeping the S/W rst > control part in resume didn't help in playing audio after resume. > So this patch adds S/W RST control part in startup function which > gets triggered for every new audio stream playback. Applied, thanks. Please use subject lines matching the subsystem.

1 0

Re: [alsa-devel] [PATCH 002/002] ASoC: Replace max98090 Device Driver
by Mark Brown 26 Jan '13

26 Jan '13

On Thu, Jan 24, 2013 at 03:28:54PM -0800, Jerry Wong wrote: > This is an odd control that is a combination of an enum and a power > management widget. The 0 value of the enum turns power off. > Not sure what is another choice. The enum value needs to be > saved so it can be restored when DAPM turns the control on. That doesn't appear to be what your control actually does - what happens if someone writes ot the control while the widget is powered down? > > +static const struct snd_kcontrol_new max98090_snd_controls[] = { > > + SOC_SINGLE("MIC Bias VCM Bandgap", M98090_REG_BIAS_CNTL, > > + M98090_VCM_MODE_SHIFT, M98090_VCM_MODE_NUM - 1, 0), > > Why is this a user visible control? > It is not controlled by DAPM. The low power function is taken care of by > /SHDN\, so this retains the user choice of bandgap or divider reference. Why would the user want this choice? > > +static int max98090_runtime_resume(struct device *dev) > > +{ > > + struct max98090_priv *max98090 = dev_get_drvdata(dev); > > + > > + regcache_cache_only(max98090->regmap, false); > > + > > + max98090_reset(max98090); > > + > > + regcache_sync(max98090->regmap); > > Do you really need to reset the device here? If anything I'd expect it > > on suspend since the reset state is probably the lowest power state. > Yes, for a soft reset where the device is not power cycled, DAPM has > the device go in and out of the low power /SHDN\ state, and the old > register values are retained. Without this, we would need to specify > that the PMIC power cycle the chip. This doesn't really answer my questions. Why would it matter if the PMIC has power cycled the chip, and if you do need to do this why is resume the time to do it?

2 4

[alsa-devel] [PATCH 04/19] oss/dmabuf: use dma_map_single
by Arnd Bergmann 25 Jan '13

25 Jan '13

The virt_to_bus/bus_to_virt functions have been deprecated for as long as I can remember, and they are used in very few remaining instances, usually in obscure ISA device drivers. The OSS sound drivers are the only ones that are still used on the ARM architecture, and only on some of the earliest StrongARM machines. The problem for converting the OSS subsystem to use dma_map_single instead is that the caller of virt_to_bus does not have a device pointer, since the subsystem has never been ported to use the common device infrastructure. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmabuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index bcc3e8e..a59c888 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap) } } dmap->raw_buf = start_addr; - dmap->raw_buf_phys = virt_to_bus(start_addr); + dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL); for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) SetPageReserved(page); @@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap) for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) ClearPageReserved(page); + dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL); free_pages((unsigned long) dmap->raw_buf, sz); dmap->raw_buf = NULL; } -- 1.8.0

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Accept 0 as a parameter block address
by Mark Brown 25 Jan '13

25 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 85 ++++++++++++++++++-------------------------- 1 file changed, 35 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bc5e383..3ba4237 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -589,27 +589,21 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].dm), be32_to_cpu(adsp1_alg[i].zm)); - if (adsp1_alg[i].dm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_DM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].dm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].dm); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp1_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP1_ZM; - region->alg = be32_to_cpu(adsp1_alg[i].alg.id); - region->base = be32_to_cpu(adsp1_alg[i].zm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_alg[i].alg.id); + region->base = be32_to_cpu(adsp1_alg[i].zm); + list_add_tail(&region->list, &dsp->alg_regions); break; case WMFW_ADSP2: @@ -623,38 +617,29 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].ym), be32_to_cpu(adsp2_alg[i].zm)); - if (adsp2_alg[i].xm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_XM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].xm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].xm); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp2_alg[i].ym) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_YM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].ym); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].ym); + list_add_tail(&region->list, &dsp->alg_regions); - if (adsp2_alg[i].zm) { - region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; - region->type = WMFW_ADSP2_ZM; - region->alg = be32_to_cpu(adsp2_alg[i].alg.id); - region->base = be32_to_cpu(adsp2_alg[i].zm); - list_add_tail(&region->list, - &dsp->alg_regions); - } + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_alg[i].alg.id); + region->base = be32_to_cpu(adsp2_alg[i].zm); + list_add_tail(&region->list, &dsp->alg_regions); break; } } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 04/19] oss/dmabuf: use dma_map_single
by Arnd Bergmann 25 Jan '13

25 Jan '13

The virt_to_bus/bus_to_virt functions have been deprecated for as long as I can remember, and they are used in very few remaining instances, usually in obscure ISA device drivers. The OSS sound drivers are the only ones that are still used on the ARM architecture, and only on some of the earliest StrongARM machines. The problem for converting the OSS subsystem to use dma_map_single instead is that the caller of virt_to_bus does not have a device pointer, since the subsystem has never been ported to use the common device infrastructure. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmabuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index bcc3e8e..a59c888 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -114,7 +114,7 @@ static int sound_alloc_dmap(struct dma_buffparms *dmap) } } dmap->raw_buf = start_addr; - dmap->raw_buf_phys = virt_to_bus(start_addr); + dmap->raw_buf_phys = dma_map_single(NULL, start_addr, dmap->buffsize, DMA_BIDIRECTIONAL); for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) SetPageReserved(page); @@ -139,6 +139,7 @@ static void sound_free_dmap(struct dma_buffparms *dmap) for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) ClearPageReserved(page); + dma_unmap_single(NULL, dmap->raw_buf_phys, dmap->buffsize, DMA_BIDIRECTIONAL); free_pages((unsigned long) dmap->raw_buf, sz); dmap->raw_buf = NULL; } -- 1.8.0

1 0

[alsa-devel] [PATCH] ASoC: arizona: Always keep SYSCLK link for DAIs
by Mark Brown 25 Jan '13

25 Jan '13

Don't disconnect SYSCLK when a DAI moves to the ASYNCCLK domain, we always need SYSCLK for operation even if it's not the reference clock for the domain. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 27b6efc..f76e982 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -978,13 +978,27 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, routes[0].sink = dai->driver->capture.stream_name; routes[1].sink = dai->driver->playback.stream_name; - routes[0].source = arizona_dai_clk_str(dai_priv->clk); - routes[1].source = arizona_dai_clk_str(dai_priv->clk); - snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + routes[0].source = arizona_dai_clk_str(dai_priv->clk); + routes[1].source = arizona_dai_clk_str(dai_priv->clk); + snd_soc_dapm_del_routes(&codec->dapm, routes, + ARRAY_SIZE(routes)); + break; + default: + break; + } - routes[0].source = arizona_dai_clk_str(clk_id); - routes[1].source = arizona_dai_clk_str(clk_id); - snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + switch (clk_id) { + case ARIZONA_CLK_ASYNCCLK: + routes[0].source = arizona_dai_clk_str(clk_id); + routes[1].source = arizona_dai_clk_str(clk_id); + snd_soc_dapm_add_routes(&codec->dapm, routes, + ARRAY_SIZE(routes)); + break; + default: + break; + } dai_priv->clk = clk_id; -- 1.7.10.4

1 0

Re: [alsa-devel] [PATCH next-20130124] Sound: pci: Fix unused variable warning in patch_sigmatel.c
by Takashi Iwai 25 Jan '13

25 Jan '13

At Fri, 25 Jan 2013 00:38:06 +0200, Stratos Karafotis wrote: > > Fix the following build warnings > > sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’: > sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable] Thanks for catching. But such temporal cast like your patch makes the code readability much worse. The code readability (thus its maintainability) is a more important factor. I'm going to fix the definition of snd_printd() itself for fixing this kind of warnings. We can use inline functions instead of empty macros to achieve the same but without triggering compiler warnings. Takashi > > Signed-off-by: Stratos Karafotis <stratosk(a)semaphore.gr> > --- > sound/pci/hda/patch_sigmatel.c | 14 +++++--------- > 1 file changed, 5 insertions(+), 9 deletions(-) > > diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c > index 0aa0ceb..f269d1f 100644 > --- a/sound/pci/hda/patch_sigmatel.c > +++ b/sound/pci/hda/patch_sigmatel.c > @@ -2365,7 +2365,6 @@ static void stac92hd71bxx_fixup_ref(struct hda_codec *codec, > static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > struct hda_jack_tbl *jack; > > if (action != HDA_FIXUP_ACT_PRE_PROBE) > @@ -2381,7 +2380,7 @@ static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, > if (jack) > jack->private_data = 0x02; > > - spec->gpio_mask |= 0x02; > + ((struct sigmatel_spec *)codec->spec)->gpio_mask |= 0x02; > > /* enable internal microphone */ > snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); > @@ -2420,19 +2419,15 @@ static void stac92hd71bxx_fixup_hp_dv5(struct hda_codec *codec, > static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > - > if (action != HDA_FIXUP_ACT_PRE_PROBE) > return; > - spec->gpio_led = 0x08; > + ((struct sigmatel_spec *)codec->spec)->gpio_led = 0x08; > } > > > static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > - struct sigmatel_spec *spec = codec->spec; > - > if (action != HDA_FIXUP_ACT_PRE_PROBE) > return; > > @@ -2456,8 +2451,9 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, > > if (find_mute_led_cfg(codec, 1)) > snd_printd("mute LED gpio %d polarity %d\n", > - spec->gpio_led, > - spec->gpio_led_polarity); > + ((struct sigmatel_spec *)codec->spec)->gpio_led, > + ((struct sigmatel_spec *)codec->spec)-> > + gpio_led_polarity); > > } > > -- > 1.8.1 >

2 2

[alsa-devel] ASoC: codec jz4740: Failed to build for use of "‘devm_ioremap_resource" on branch "for-next"
by Lin, Mengdong 25 Jan '13

25 Jan '13

Hi, I failed to build the branch “for-next”. Here is error information: sound/soc/codecs/jz4740.c: In function ‘jz4740_codec_probe’: sound/soc/codecs/jz4740.c:364:2: error: implicit declaration of function ‘devm_ioremap_resource’ [-Werror=implicit-function-declaration] sound/soc/codecs/jz4740.c:364:7: warning: assignment makes pointer from integer without a cast [enabled by default] It seems the newly introduced “devm_ioremap_resource” is to replace devm_request_and_ioremap(). But its definition are absent from the kernel. I cannot find where it’s defined, neither in device.h nor in io.h . Thanks Mengdong

1 0

[alsa-devel] hdsp rpm support
by Roman Haefeli 25 Jan '13

25 Jan '13

Hi everyone I would like to know current state of the RME HDSP RPM support. There was a thread [1] in 2008 on this list about the issue and things looked promising back then. However, I'm still unable to upload the firmware with alsa-driver from git on my Ubuntu 11.04 (32bit) machine with a 2.6.38 kernel. I posted a detailed description of the issue on the alsa-user list here [2]. Is it supposed to work with Cardbus? If not, is there something I can do to help support it? Any hints appreciated. Roman [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-May/008128.html [2] http://article.gmane.org/gmane.linux.alsa.user/36022

5 12

[alsa-devel] [PATCH 00/13] A few more fixes / enhancements for HD-audio
by Takashi Iwai 24 Jan '13

24 Jan '13

Not surprisingly, there are yet a few more patches for HD-audio. All patches are found in for-next branch of sound git tree as usual. Takashi

1 13

Re: [alsa-devel] spdif input, source synchronous?
by Rafal Powierski 24 Jan '13

24 Jan '13

Dave, it took me some time, but I finally got a proper equipment to test ALC886 SPDIF input. (hw:1,1,0) The test results show that the input is not "source synchronous" experiment: Analog->digital (spdif,toslink) Audio converter box, where internal crystal is removed and replaced with precision adjustable PLL-based gen. 2 simple independent sinus audio generators, providing audio source for ADC L,R inputs software: stream consequent samples (L,R) are compared. frequency of identical consequent samples (32bits) is calculated (ppm) results: crystal marking is: 12288000 = 256*48kHz gen. Hz | gen. ppm | duplicate probability (ppm) 12288000 0 0 12287856 -11.72 2 12287840 -13.02 4 12287824 -14.32 5 12287808 -15.62 7 12287792 -16.93 9 12287776 -18.22 10 12287760 -19.53 13 12287744 -20.83 14 12287728 -22.13 16 12287712 -23.44 17 12287696 -24.74 18 12287680 -26.04 20 12287664 -27.34 21 It seams that my HDA clock is a bit slower then definition ( about -10ppm) Therefore resampling starts below -10ppm It probably means that when spdif signal is 0ppm, about 10ppm of samples are missing. Is this a bug? or it is planed like this? Is this a software problem or hardware problem (codec or DMA sequencer) or configuration problem? since it is not a real resampling (no signal reconstruction takes place) samples are just repeated or skipped, it seams not right to me. Who could comment on this? Regards, Rafal -----Original Message----- From: Dave Platt Sent: Monday, November 12, 2012 11:40 PM To: powierski(a)mediascore.de Subject: re: [alsa-devel] spdif input, source synchronous? > Does anyone know how the SPDIF input is working on HDA? > Is it using a source-synchronous data transfer over HDA? > In my arrangement (ALC886) I can set up any rate and I am receiving a > rectangular “resampling”. > – is this a bug? > will this stay? so I can use higher ratio and some kind of software PLL to > recover input rate? I tested out S/PDIF input on my ALC262 setup some time ago. What I found, at the time, was that the data capture did appear to be driven by the rate of the incoming signal. My outboard ADC provides a 44100 sample/second rate, and that seemed to be what I was getting. I tried forcing the capture to 48000 samples per second, and saw no difference in the captured file size... the number of samples per second actually captured seemed to be the same (within experimental error) and I didn't see any evidence of resampling going on. Now, this was a capture with "hw:Intel,1" as the device. I expect that I would have seen resampling, and a different type of delivery if I had specified "plughw:Intel,1" or some other input style which had gone through the plug interface. Unfortunately I don't have any way of generating a CD-rate data stream with a specific set of known values... I'd need a CD player with an S/PDIF output and I don't have one of those handy. If I can figure out a way of doing this I'll try generating a set of CDs with a known-value pattern, play it, capture it, and see if it does or does not exactly correlate.

2 1

[alsa-devel] Soc MC13783: Improve driver to handle LineIn
by Gaëtan Carlier 24 Jan '13

24 Jan '13

Hello, I would like to improve MC13783 driver to be able to control bypass of LineIn (RXINR/L) and feed Analog LineIn stream into ouput Mixer via ADDRXIN flag. This first step one is for test purpose but I also need to select RXINR/L as capture source (I saw in RM that CODEC must be enabled and only RXINR flag must be selected as input) but nothing seems to be captured this is why I need bypass working to test input level, and so on. I try to create a Mixer based on other driver but it doesn't work. I try to finc information about widget macro SND_SOC_DAPM_* but it is hard to find something relevant. * Which widgets can be link together ? * Is there functional difference between VIRT and non-VIRT widget ? I will post my code in some minutes after clean-up. Thanks a lot. Gaëtan Carlier.

1 1

[alsa-devel] [PATCH Resend 0/4] ASoC: tegra: Some fixes in codec files
by Sachin Kamat 24 Jan '13

24 Jan '13

Resending this series with Stephen Warren's 'Reviewed-by' tag and dropping a patch as suggested by him. Compile tested using tegra_defconfig on the for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Sachin Kamat (4): ASoC: tegra_wm9712: Remove __devinitconst attribute ASoC: tegra20_ac97: Remove __devinitconst attribute ASoC: tegra: Use NULL instead of 0 for pointers ASoC: tegra: Staticize some functions in tegra30_i2s.c sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) -- 1.7.4.1

2 5

[alsa-devel] [PATCH 1/3] ASoC: soc-compress: Add missing brackets around else
by Charles Keepax 24 Jan '13

24 Jan '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 5fbfb06..80040f0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -112,10 +112,11 @@ static int soc_compr_free(struct snd_compr_stream *cstream) snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); - } else + } else { rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); + } } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, -- 1.7.2.5

3 4

[alsa-devel] [PATCH V2 0/2] ARM/ASoC: Davinci: Device Tree Update
by Hebbar Gururaja 24 Jan '13

24 Jan '13

1. Add pinctrl for McASP Module 2. Add DT support for Davinci machine platform This patch-set is tested on Davinci platform (DA850 EVM). This series applies on top of tag next-20130103 git tree https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git These patches cannot be tested directly on mainline kernel (DT based). This is because dma is not yet completely ported to Davinci ASoC. I had to add few hacks to get Audio working on Davinci DA850 platform. For reference, I have placed the entire working tree @ Tree : https://gitorious.org/davinci-asoc-dt/davinci-asoc-dt Branch : dev_next-20130103_davinci_asoc_dt Changes in V2 - Remove reference to Linux & software details from DT binding Hebbar, Gururaja (2): ASoC: davinci-mcasp: Add pinctrl support ASoC: Davinci: machine: Add device tree binding .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ sound/soc/davinci/davinci-evm.c | 179 +++++++++++++++++--- sound/soc/davinci/davinci-mcasp.c | 7 + 3 files changed, 219 insertions(+), 20 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

3 13

Re: [alsa-devel] [PATCH V6 01/10] ASoC: Samsung: Rename samsung i2s secondary device name
by Mark Brown 24 Jan '13

24 Jan '13

On Fri, Jan 18, 2013 at 05:17:00PM +0530, Padmavathi Venna wrote: > All Samsung SoCs has max 3 i2s controllers. So the i2s secondary fifo > interface device id was named as samsung-i2s.4. Renaming this to > "samsung-i2s-sec" to support device tree in i2s driver. Applied, thanks.

1 0

[alsa-devel] [PATCH 002/002] ASoC: Replace max98090 Device Driver
by Jerry Wong 23 Jan '13

23 Jan '13

From: Jerry Wong <jerry.wong(a)maximintegrated.com> This patch completes the replacement of the existing max98090 driver, by installing a more complete driver. Signed-off-by: Jerry Wong <jerry.wong(a)maximintegrated.com> Tested-by: Matthew Mowdy <matthew.mowdy(a)maximintegrated.com> Reviewed-by: Ralph Birt <ralph.birt(a)maximintegrated.com> --- I have updated most per the review comments, except... "External MIC Mux" is a mux with 0=power down, and exists in DAPM as well "Digital Sidetone Source", with a master SHDN has no power impact, so was left out of DAPM "Quick Setup System Clock" was just a question what they do... they override the clock settings for common use cases. "VCM bandgap reference", with a master SHDN has no power impact, so was left out of DAPM include/sound/max98090.h | 29 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98090.c | 2603 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98090.h | 1537 +++++++++++++++++++++++++ 5 files changed, 4175 insertions(+), 0 deletions(-) create mode 100755 include/sound/max98090.h create mode 100755 sound/soc/codecs/max98090.c create mode 100755 sound/soc/codecs/max98090.h diff --git a/include/sound/max98090.h b/include/sound/max98090.h new file mode 100755 index 0000000..95efb13 --- /dev/null +++ b/include/sound/max98090.h @@ -0,0 +1,29 @@ +/* + * Platform data for MAX98090 + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SOUND_MAX98090_PDATA_H__ +#define __SOUND_MAX98090_PDATA_H__ + +/* codec platform data */ +struct max98090_pdata { + + /* Analog/digital microphone configuration: + * 0 = analog microphone input (normal setting) + * 1 = digital microphone input + */ + unsigned int digmic_left_mode:1; + unsigned int digmic_right_mode:1; + unsigned int digmic_3_mode:1; + unsigned int digmic_4_mode:1; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0e368d4..3a84782 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -44,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9768 if I2C @@ -267,6 +268,9 @@ config SND_SOC_LM49453 config SND_SOC_MAX98088 tristate +config SND_SOC_MAX98090 + tristate + config SND_SOC_MAX98095 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aa56312..f6e8e36 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -34,6 +34,7 @@ snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o +snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o @@ -157,6 +158,7 @@ obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX98090) += snd-soc-max98090.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c new file mode 100755 index 0000000..b796c55 --- /dev/null +++ b/sound/soc/codecs/max98090.c @@ -0,0 +1,2603 @@ +/* + * max98090.c -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/max98090.h> +#include "max98090.h" + +#include <linux/version.h> + +#define DEBUG +#define EXTMIC_METHOD +#define EXTMIC_METHOD_TEST + +/* Allows for sparsely populated register maps */ +static struct reg_default max98090_reg[] = { + { 0x00, 0x00 }, /* 00 Software Reset */ + { 0x03, 0x04 }, /* 03 Interrupt Masks */ + { 0x04, 0x00 }, /* 04 System Clock Quick */ + { 0x05, 0x00 }, /* 05 Sample Rate Quick */ + { 0x06, 0x00 }, /* 06 DAI Interface Quick */ + { 0x07, 0x00 }, /* 07 DAC Path Quick */ + { 0x08, 0x00 }, /* 08 Mic/Direct to ADC Quick */ + { 0x09, 0x00 }, /* 09 Line to ADC Quick */ + { 0x0A, 0x00 }, /* 0A Analog Mic Loop Quick */ + { 0x0B, 0x00 }, /* 0B Analog Line Loop Quick */ + { 0x0C, 0x00 }, /* 0C Reserved */ + { 0x0D, 0x00 }, /* 0D Input Config */ + { 0x0E, 0x1B }, /* 0E Line Input Level */ + { 0x0F, 0x00 }, /* 0F Line Config */ + + { 0x10, 0x14 }, /* 10 Mic1 Input Level */ + { 0x11, 0x14 }, /* 11 Mic2 Input Level */ + { 0x12, 0x00 }, /* 12 Mic Bias Voltage */ + { 0x13, 0x00 }, /* 13 Digital Mic Config */ + { 0x14, 0x00 }, /* 14 Digital Mic Mode */ + { 0x15, 0x00 }, /* 15 Left ADC Mixer */ + { 0x16, 0x00 }, /* 16 Right ADC Mixer */ + { 0x17, 0x03 }, /* 17 Left ADC Level */ + { 0x18, 0x03 }, /* 18 Right ADC Level */ + { 0x19, 0x00 }, /* 19 ADC Biquad Level */ + { 0x1A, 0x00 }, /* 1A ADC Sidetone */ + { 0x1B, 0x00 }, /* 1B System Clock */ + { 0x1C, 0x00 }, /* 1C Clock Mode */ + { 0x1D, 0x00 }, /* 1D Any Clock 1 */ + { 0x1E, 0x00 }, /* 1E Any Clock 2 */ + { 0x1F, 0x00 }, /* 1F Any Clock 3 */ + + { 0x20, 0x00 }, /* 20 Any Clock 4 */ + { 0x21, 0x00 }, /* 21 Master Mode */ + { 0x22, 0x00 }, /* 22 Interface Format */ + { 0x23, 0x00 }, /* 23 TDM Format 1*/ + { 0x24, 0x00 }, /* 24 TDM Format 2*/ + { 0x25, 0x00 }, /* 25 I/O Configuration */ + { 0x26, 0x80 }, /* 26 Filter Config */ + { 0x27, 0x00 }, /* 27 DAI Playback Level */ + { 0x28, 0x00 }, /* 28 EQ Playback Level */ + { 0x29, 0x00 }, /* 29 Left HP Mixer */ + { 0x2A, 0x00 }, /* 2A Right HP Mixer */ + { 0x2B, 0x00 }, /* 2B HP Control */ + { 0x2C, 0x1A }, /* 2C Left HP Volume */ + { 0x2D, 0x1A }, /* 2D Right HP Volume */ + { 0x2E, 0x00 }, /* 2E Left Spk Mixer */ + { 0x2F, 0x00 }, /* 2F Right Spk Mixer */ + + { 0x30, 0x00 }, /* 30 Spk Control */ + { 0x31, 0x2C }, /* 31 Left Spk Volume */ + { 0x32, 0x2C }, /* 32 Right Spk Volume */ + { 0x33, 0x00 }, /* 33 ALC Timing */ + { 0x34, 0x00 }, /* 34 ALC Compressor */ + { 0x35, 0x00 }, /* 35 ALC Expander */ + { 0x36, 0x00 }, /* 36 ALC Gain */ + { 0x37, 0x00 }, /* 37 Rcv/Line OutL Mixer */ + { 0x38, 0x00 }, /* 38 Rcv/Line OutL Control */ + { 0x39, 0x15 }, /* 39 Rcv/Line OutL Volume */ + { 0x3A, 0x00 }, /* 3A Line OutR Mixer */ + { 0x3B, 0x00 }, /* 3B Line OutR Control */ + { 0x3C, 0x15 }, /* 3C Line OutR Volume */ + { 0x3D, 0x00 }, /* 3D Jack Detect */ + { 0x3E, 0x00 }, /* 3E Input Enable */ + { 0x3F, 0x00 }, /* 3F Output Enable */ + + { 0x40, 0x00 }, /* 40 Level Control */ + { 0x41, 0x00 }, /* 41 DSP Filter Enable */ + { 0x42, 0x00 }, /* 42 Bias Control */ + { 0x43, 0x00 }, /* 43 DAC Control */ + { 0x44, 0x06 }, /* 44 ADC Control */ + { 0x45, 0x00 }, /* 45 Device Shutdown */ + { 0x46, 0x00 }, /* 46 Equalizer Band 1 Coefficient B0 */ + { 0x47, 0x00 }, /* 47 Equalizer Band 1 Coefficient B0 */ + { 0x48, 0x00 }, /* 48 Equalizer Band 1 Coefficient B0 */ + { 0x49, 0x00 }, /* 49 Equalizer Band 1 Coefficient B1 */ + { 0x4A, 0x00 }, /* 4A Equalizer Band 1 Coefficient B1 */ + { 0x4B, 0x00 }, /* 4B Equalizer Band 1 Coefficient B1 */ + { 0x4C, 0x00 }, /* 4C Equalizer Band 1 Coefficient B2 */ + { 0x4D, 0x00 }, /* 4D Equalizer Band 1 Coefficient B2 */ + { 0x4E, 0x00 }, /* 4E Equalizer Band 1 Coefficient B2 */ + { 0x4F, 0x00 }, /* 4F Equalizer Band 1 Coefficient A1 */ + + { 0x50, 0x00 }, /* 50 Equalizer Band 1 Coefficient A1 */ + { 0x51, 0x00 }, /* 51 Equalizer Band 1 Coefficient A1 */ + { 0x52, 0x00 }, /* 52 Equalizer Band 1 Coefficient A2 */ + { 0x53, 0x00 }, /* 53 Equalizer Band 1 Coefficient A2 */ + { 0x54, 0x00 }, /* 54 Equalizer Band 1 Coefficient A2 */ + { 0x55, 0x00 }, /* 55 Equalizer Band 2 Coefficient B0 */ + { 0x56, 0x00 }, /* 56 Equalizer Band 2 Coefficient B0 */ + { 0x57, 0x00 }, /* 57 Equalizer Band 2 Coefficient B0 */ + { 0x58, 0x00 }, /* 58 Equalizer Band 2 Coefficient B1 */ + { 0x59, 0x00 }, /* 59 Equalizer Band 2 Coefficient B1 */ + { 0x5A, 0x00 }, /* 5A Equalizer Band 2 Coefficient B1 */ + { 0x5B, 0x00 }, /* 5B Equalizer Band 2 Coefficient B2 */ + { 0x5C, 0x00 }, /* 5C Equalizer Band 2 Coefficient B2 */ + { 0x5D, 0x00 }, /* 5D Equalizer Band 2 Coefficient B2 */ + { 0x5E, 0x00 }, /* 5E Equalizer Band 2 Coefficient A1 */ + { 0x5F, 0x00 }, /* 5F Equalizer Band 2 Coefficient A1 */ + + { 0x60, 0x00 }, /* 60 Equalizer Band 2 Coefficient A1 */ + { 0x61, 0x00 }, /* 61 Equalizer Band 2 Coefficient A2 */ + { 0x62, 0x00 }, /* 62 Equalizer Band 2 Coefficient A2 */ + { 0x63, 0x00 }, /* 63 Equalizer Band 2 Coefficient A2 */ + { 0x64, 0x00 }, /* 64 Equalizer Band 3 Coefficient B0 */ + { 0x65, 0x00 }, /* 65 Equalizer Band 3 Coefficient B0 */ + { 0x66, 0x00 }, /* 66 Equalizer Band 3 Coefficient B0 */ + { 0x67, 0x00 }, /* 67 Equalizer Band 3 Coefficient B1 */ + { 0x68, 0x00 }, /* 68 Equalizer Band 3 Coefficient B1 */ + { 0x69, 0x00 }, /* 69 Equalizer Band 3 Coefficient B1 */ + { 0x6A, 0x00 }, /* 6A Equalizer Band 3 Coefficient B2 */ + { 0x6B, 0x00 }, /* 6B Equalizer Band 3 Coefficient B2 */ + { 0x6C, 0x00 }, /* 6C Equalizer Band 3 Coefficient B2 */ + { 0x6D, 0x00 }, /* 6D Equalizer Band 3 Coefficient A1 */ + { 0x6E, 0x00 }, /* 6E Equalizer Band 3 Coefficient A1 */ + { 0x6F, 0x00 }, /* 6F Equalizer Band 3 Coefficient A1 */ + + { 0x70, 0x00 }, /* 70 Equalizer Band 3 Coefficient A2 */ + { 0x71, 0x00 }, /* 71 Equalizer Band 3 Coefficient A2 */ + { 0x72, 0x00 }, /* 72 Equalizer Band 3 Coefficient A2 */ + { 0x73, 0x00 }, /* 73 Equalizer Band 4 Coefficient B0 */ + { 0x74, 0x00 }, /* 74 Equalizer Band 4 Coefficient B0 */ + { 0x75, 0x00 }, /* 75 Equalizer Band 4 Coefficient B0 */ + { 0x76, 0x00 }, /* 76 Equalizer Band 4 Coefficient B1 */ + { 0x77, 0x00 }, /* 77 Equalizer Band 4 Coefficient B1 */ + { 0x78, 0x00 }, /* 78 Equalizer Band 4 Coefficient B1 */ + { 0x79, 0x00 }, /* 79 Equalizer Band 4 Coefficient B2 */ + { 0x7A, 0x00 }, /* 7A Equalizer Band 4 Coefficient B2 */ + { 0x7B, 0x00 }, /* 7B Equalizer Band 4 Coefficient B2 */ + { 0x7C, 0x00 }, /* 7C Equalizer Band 4 Coefficient A1 */ + { 0x7D, 0x00 }, /* 7D Equalizer Band 4 Coefficient A1 */ + { 0x7E, 0x00 }, /* 7E Equalizer Band 4 Coefficient A1 */ + { 0x7F, 0x00 }, /* 7F Equalizer Band 4 Coefficient A2 */ + + { 0x80, 0x00 }, /* 80 Equalizer Band 4 Coefficient A2 */ + { 0x81, 0x00 }, /* 81 Equalizer Band 4 Coefficient A2 */ + { 0x82, 0x00 }, /* 82 Equalizer Band 5 Coefficient B0 */ + { 0x83, 0x00 }, /* 83 Equalizer Band 5 Coefficient B0 */ + { 0x84, 0x00 }, /* 84 Equalizer Band 5 Coefficient B0 */ + { 0x85, 0x00 }, /* 85 Equalizer Band 5 Coefficient B1 */ + { 0x86, 0x00 }, /* 86 Equalizer Band 5 Coefficient B1 */ + { 0x87, 0x00 }, /* 87 Equalizer Band 5 Coefficient B1 */ + { 0x88, 0x00 }, /* 88 Equalizer Band 5 Coefficient B2 */ + { 0x89, 0x00 }, /* 89 Equalizer Band 5 Coefficient B2 */ + { 0x8A, 0x00 }, /* 8A Equalizer Band 5 Coefficient B2 */ + { 0x8B, 0x00 }, /* 8B Equalizer Band 5 Coefficient A1 */ + { 0x8C, 0x00 }, /* 8C Equalizer Band 5 Coefficient A1 */ + { 0x8D, 0x00 }, /* 8D Equalizer Band 5 Coefficient A1 */ + { 0x8E, 0x00 }, /* 8E Equalizer Band 5 Coefficient A2 */ + { 0x8F, 0x00 }, /* 8F Equalizer Band 5 Coefficient A2 */ + + { 0x90, 0x00 }, /* 90 Equalizer Band 5 Coefficient A2 */ + { 0x91, 0x00 }, /* 91 Equalizer Band 6 Coefficient B0 */ + { 0x92, 0x00 }, /* 92 Equalizer Band 6 Coefficient B0 */ + { 0x93, 0x00 }, /* 93 Equalizer Band 6 Coefficient B0 */ + { 0x94, 0x00 }, /* 94 Equalizer Band 6 Coefficient B1 */ + { 0x95, 0x00 }, /* 95 Equalizer Band 6 Coefficient B1 */ + { 0x96, 0x00 }, /* 96 Equalizer Band 6 Coefficient B1 */ + { 0x97, 0x00 }, /* 97 Equalizer Band 6 Coefficient B2 */ + { 0x98, 0x00 }, /* 98 Equalizer Band 6 Coefficient B2 */ + { 0x99, 0x00 }, /* 99 Equalizer Band 6 Coefficient B2 */ + { 0x9A, 0x00 }, /* 9A Equalizer Band 6 Coefficient A1 */ + { 0x9B, 0x00 }, /* 9B Equalizer Band 6 Coefficient A1 */ + { 0x9C, 0x00 }, /* 9C Equalizer Band 6 Coefficient A1 */ + { 0x9D, 0x00 }, /* 9D Equalizer Band 6 Coefficient A2 */ + { 0x9E, 0x00 }, /* 9E Equalizer Band 6 Coefficient A2 */ + { 0x9F, 0x00 }, /* 9F Equalizer Band 6 Coefficient A2 */ + + { 0xA0, 0x00 }, /* A0 Equalizer Band 7 Coefficient B0 */ + { 0xA1, 0x00 }, /* A1 Equalizer Band 7 Coefficient B0 */ + { 0xA2, 0x00 }, /* A2 Equalizer Band 7 Coefficient B0 */ + { 0xA3, 0x00 }, /* A3 Equalizer Band 7 Coefficient B1 */ + { 0xA4, 0x00 }, /* A4 Equalizer Band 7 Coefficient B1 */ + { 0xA5, 0x00 }, /* A5 Equalizer Band 7 Coefficient B1 */ + { 0xA6, 0x00 }, /* A6 Equalizer Band 7 Coefficient B2 */ + { 0xA7, 0x00 }, /* A7 Equalizer Band 7 Coefficient B2 */ + { 0xA8, 0x00 }, /* A8 Equalizer Band 7 Coefficient B2 */ + { 0xA9, 0x00 }, /* A9 Equalizer Band 7 Coefficient A1 */ + { 0xAA, 0x00 }, /* AA Equalizer Band 7 Coefficient A1 */ + { 0xAB, 0x00 }, /* AB Equalizer Band 7 Coefficient A1 */ + { 0xAC, 0x00 }, /* AC Equalizer Band 7 Coefficient A2 */ + { 0xAD, 0x00 }, /* AD Equalizer Band 7 Coefficient A2 */ + { 0xAE, 0x00 }, /* AE Equalizer Band 7 Coefficient A2 */ + { 0xAF, 0x00 }, /* AF ADC Biquad Coefficient B0 */ + + { 0xB0, 0x00 }, /* B0 ADC Biquad Coefficient B0 */ + { 0xB1, 0x00 }, /* B1 ADC Biquad Coefficient B0 */ + { 0xB2, 0x00 }, /* B2 ADC Biquad Coefficient B1 */ + { 0xB3, 0x00 }, /* B3 ADC Biquad Coefficient B1 */ + { 0xB4, 0x00 }, /* B4 ADC Biquad Coefficient B1 */ + { 0xB5, 0x00 }, /* B5 ADC Biquad Coefficient B2 */ + { 0xB6, 0x00 }, /* B6 ADC Biquad Coefficient B2 */ + { 0xB7, 0x00 }, /* B7 ADC Biquad Coefficient B2 */ + { 0xB8, 0x00 }, /* B8 ADC Biquad Coefficient A1 */ + { 0xB9, 0x00 }, /* B9 ADC Biquad Coefficient A1 */ + { 0xBA, 0x00 }, /* BA ADC Biquad Coefficient A1 */ + { 0xBB, 0x00 }, /* BB ADC Biquad Coefficient A2 */ + { 0xBC, 0x00 }, /* BC ADC Biquad Coefficient A2 */ + { 0xBD, 0x00 }, /* BD ADC Biquad Coefficient A2 */ + { 0xBE, 0x00 }, /* BE Digital Mic 3 Volume */ + { 0xBF, 0x00 }, /* BF Digital Mic 4 Volume */ + + { 0xC0, 0x00 }, /* C0 Digital Mic 34 Biquad Pre Atten */ + { 0xC1, 0x00 }, /* C1 Record TDM Slot */ + { 0xC2, 0x00 }, /* C2 Sample Rate */ + { 0xC3, 0x00 }, /* C3 Digital Mic 34 Biquad Coefficient C3 */ + { 0xC4, 0x00 }, /* C4 Digital Mic 34 Biquad Coefficient C4 */ + { 0xC5, 0x00 }, /* C5 Digital Mic 34 Biquad Coefficient C5 */ + { 0xC6, 0x00 }, /* C6 Digital Mic 34 Biquad Coefficient C6 */ + { 0xC7, 0x00 }, /* C7 Digital Mic 34 Biquad Coefficient C7 */ + { 0xC8, 0x00 }, /* C8 Digital Mic 34 Biquad Coefficient C8 */ + { 0xC9, 0x00 }, /* C9 Digital Mic 34 Biquad Coefficient C9 */ + { 0xCA, 0x00 }, /* CA Digital Mic 34 Biquad Coefficient CA */ + { 0xCB, 0x00 }, /* CB Digital Mic 34 Biquad Coefficient CB */ + { 0xCC, 0x00 }, /* CC Digital Mic 34 Biquad Coefficient CC */ + { 0xCD, 0x00 }, /* CD Digital Mic 34 Biquad Coefficient CD */ + { 0xCE, 0x00 }, /* CE Digital Mic 34 Biquad Coefficient CE */ + { 0xCF, 0x00 }, /* CF Digital Mic 34 Biquad Coefficient CF */ + + { 0xD0, 0x00 }, /* D0 Digital Mic 34 Biquad Coefficient D0 */ + { 0xD1, 0x00 }, /* D1 Digital Mic 34 Biquad Coefficient D1 */ +}; + +static bool max98090_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_IRQ_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_REV_ID: + return true; + default: + return false; + } +} + +static bool max98090_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98090_REG_IRQ_STATUS: + case M98090_REG_JACK_STATUS: + case M98090_REG_IRQ_ENABLE: + case M98090_REG_RESERVED: + case M98090_REG_CFG_INPUT: + case M98090_REG_LVL_LINE: + case M98090_REG_CFG_LINE: + case M98090_REG_LVL_MIC1: + case M98090_REG_LVL_MIC2: + case M98090_REG_MIC_BIAS: + case M98090_REG_MIC_CFG1: + case M98090_REG_MIC_CFG2: + case M98090_REG_MIX_ADC_L: + case M98090_REG_MIX_ADC_R: + case M98090_REG_LVL_ADC_L: + case M98090_REG_LVL_ADC_R: + case M98090_REG_LVL_BIQUAD: + case M98090_REG_LVL_SIDETONE: + case M98090_REG_SYS_CLOCK: + case M98090_REG_CLOCK_MODE: + case M98090_REG_CLOCK_DAI1_NI_HI: + case M98090_REG_CLOCK_DAI2_NI_LO: + case M98090_REG_CLOCK_DAI3_MI_HI: + case M98090_REG_CLOCK_DAI4_MI_LO: + case M98090_REG_CLOCK_MAS_MODE: + case M98090_REG_DAI_INTERFACE_FORMAT: + case M98090_REG_DAI_TDM_CONTROL: + case M98090_REG_DAI_TDM_FORMAT: + case M98090_REG_DAI_IOCFG: + case M98090_REG_DAI_FILTERS: + case M98090_REG_DAI_LVL: + case M98090_REG_DAI_LVL_EQ: + case M98090_REG_MIX_HP_LEFT: + case M98090_REG_MIX_HP_RIGHT: + case M98090_REG_MIX_HP_CNTL: + case M98090_REG_LVL_HP_LEFT: + case M98090_REG_LVL_HP_RIGHT: + case M98090_REG_MIX_SPK_LEFT: + case M98090_REG_MIX_SPK_RIGHT: + case M98090_REG_MIX_SPK_CNTL: + case M98090_REG_LVL_SPK_LEFT: + case M98090_REG_LVL_SPK_RIGHT: + case M98090_REG_ALC_TIMING: + case M98090_REG_ALC_CMPR: + case M98090_REG_ALC_EXP: + case M98090_REG_LVL_ALC: + case M98090_REG_MIX_RCV_LEFT: + case M98090_REG_MIX_RCV_CTRL_LEFT: + case M98090_REG_LVL_RCV_LEFT: + case M98090_REG_MIX_RCV_RIGHT: + case M98090_REG_MIX_RCV_CNTL_RIGHT: + case M98090_REG_LVL_RCV_RIGHT: + case M98090_REG_CFG_JACK: + case M98090_REG_PWR_EN_IN: + case M98090_REG_PWR_EN_OUT: + case M98090_REG_CFG_LVL: + case M98090_REG_DSP_EQ_EN: + case M98090_REG_BIAS_CNTL: + case M98090_REG_DAC_CFG: + case M98090_REG_ADC_CFG: + case M98090_REG_PWR_SYS: + case M98090_REG_EQ_BASE ... M98090_REG_EQ_BASE + 0x68: + case M98090_REG_BIQUAD_BASE ... M98090_REG_BIQUAD_BASE + 0x0E: + case M98090_REG_DMIC3_VOLUME: + case M98090_REG_DMIC4_VOLUME: + case M98090_REG_DMIC34_BQ_PREATTEN: + case M98090_REG_RECORD_TDM_SLOT: + case M98090_REG_SAMPLE_RATE: + case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + return true; + default: + return false; + } +} + +static int max98090_reset(struct max98090_priv *max98090) +{ + int ret; + + /* Reset the codec by writing to this write-only reset register */ + ret = regmap_write(max98090->regmap, M98090_REG_SW_RESET, + M98090_SWRESET_MASK); + if (ret < 0) { + dev_err(max98090->codec->dev, + "Failed to reset codec: %d\n", ret); + return ret; + } + + msleep(20); + return ret; +} + +static const unsigned int max98090_micboost_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_mic_tlv, 0, 100, 0); + +static const unsigned int max98090_lin_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-600, 300, 0), + 4, 5, TLV_DB_SCALE_ITEM(1400, 600, 0), +}; + +static const DECLARE_TLV_DB_SCALE(max98090_adcboost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_adc_tlv, -1200, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(max98090_dv_tlv, -1500, 100, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_sidetone_tlv, -6050, 200, 0); + +static const DECLARE_TLV_DB_SCALE(max98090_alc_tlv, -1500, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcmakeup_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alccomp_tlv, -3100, 100, 0); +static const DECLARE_TLV_DB_SCALE(max98090_alcexp_tlv, -6600, 100, 0); + +static const unsigned int max98090_mixout_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(-1200, 250, 0), + 2, 3, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +static const unsigned int max98090_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98090_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 4, TLV_DB_SCALE_ITEM(-4800, 400, 0), + 5, 10, TLV_DB_SCALE_ITEM(-2900, 300, 0), + 11, 14, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 15, 29, TLV_DB_SCALE_ITEM(-500, 100, 0), + 30, 39, TLV_DB_SCALE_ITEM(950, 50, 0), +}; + +static const unsigned int max98090_rcv_lout_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + +static int max98090_extmic_mux_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, M98090_REG_CFG_LINE); + + val = (val & M98090_EXTMIC_MASK) >> M98090_EXTMIC_SHIFT; + + max98090->extmic_mux = sel; + + /* Setting a channel is only valid if it is already On */ + if ((val == 1) || (val == 2)) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, M98090_REG_CFG_LINE, + M98090_EXTMIC_MASK, + sel << M98090_EXTMIC_SHIFT); + + return 0; +} + +static int max98090_extmic_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned int val = snd_soc_read(codec, M98090_REG_CFG_LINE); + + val = (val & M98090_EXTMIC_MASK) >> M98090_EXTMIC_SHIFT; + + if ((val == 1) || (val == 2)) { + /* If on, return the channel */ + val = val - 1; + max98090->extmic_mux = val; + } else { + /* If off, return last stored value */ + val = max98090->extmic_mux; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_get_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_LVL_MIC1: + select = &(max98090->mic1pre); + break; + case M98090_REG_LVL_MIC2: + select = &(max98090->mic2pre); + break; + case M98090_REG_LVL_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + if (val >= 1) { + /* If on, return the volume */ + val = val - 1; + *select = val; + } else { + /* If off, return last stored value */ + val = *select; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int mask = (1 << fls(mc->max)) - 1; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int val = snd_soc_read(codec, mc->reg); + unsigned int *select; + + switch (mc->reg) { + case M98090_REG_LVL_MIC1: + select = &(max98090->mic1pre); + break; + case M98090_REG_LVL_MIC2: + select = &(max98090->mic2pre); + break; + case M98090_REG_LVL_SIDETONE: + select = &(max98090->sidetone); + break; + default: + return -EINVAL; + } + + val = (val >> mc->shift) & mask; + + *select = sel; + + /* Setting a volume is only valid if it is already On */ + if (val >= 1) { + sel = sel + 1; + } else { + /* Write what was already there */ + sel = val; + } + + snd_soc_update_bits(codec, mc->reg, + mask << mc->shift, + sel << mc->shift); + + return 0; +} + +static const char * max98090_extmic_mux_text[] = { "MIC1", "MIC2" }; + +static const struct soc_enum max98090_extmic_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC_SHIFT, + ARRAY_SIZE(max98090_extmic_mux_text), + max98090_extmic_mux_text); + +static const char * max98090_mixg_text[] = { "Normal", "6dB" }; + +static const struct soc_enum max98090_mixg135_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_LINE, M98090_MIXG135_SHIFT, + ARRAY_SIZE(max98090_mixg_text), max98090_mixg_text); + +static const struct soc_enum max98090_mixg246_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_LINE, M98090_MIXT246_SHIFT, + ARRAY_SIZE(max98090_mixg_text), max98090_mixg_text); + +static const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; + +static const struct soc_enum max98090_osr128_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CFG, M98090_ADC_OSR128_SHIFT, + ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); + +static const char * max98090_enableddisabled_text[] = + { "Disabled", "Enabled" }; +static const char * max98090_enableddisabled_inv_text[] = + { "Enabled", "Disabled" }; +/* Note: Inverted Logic (0 = Enabled) */ + +static const struct soc_enum max98090_eqclpn_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_LVL_EQ, M98090_DAI_LVL_EQCLPN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct soc_enum max98090_sdien_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_SDIEN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_text), + max98090_enableddisabled_text); + +static const struct soc_enum max98090_sdoen_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_SDOEN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_text), + max98090_enableddisabled_text); + +static const struct soc_enum max98090_hizoff_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_HIZOFF_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const char *max98090_fltr_mode_text[] = { "Voice", "Music" }; + +static const struct soc_enum max98090_filter_mode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_FILTERS, M98090_DAI_FLT_MODE_SHIFT, + ARRAY_SIZE(max98090_fltr_mode_text), max98090_fltr_mode_text); + +static const struct soc_enum max98090_filter_dmic34mode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DMIC34MODE_SHIFT, + ARRAY_SIZE(max98090_fltr_mode_text), max98090_fltr_mode_text); + +static const char * max98090_dsts_text[] = + { "None", "Left", "Right", "Left and Right" }; + +static const struct soc_enum max98090_dsts_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_SIDETONE, M98090_DSTS_SHIFT, + ARRAY_SIZE(max98090_dsts_text), max98090_dsts_text); + +static const char * max98090_alcatk_text[] = + { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; + +static const struct soc_enum max98090_alcatk_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_TIMING, M98090_ALCATK_SHIFT, + ARRAY_SIZE(max98090_alcatk_text), max98090_alcatk_text); + +static const char * max98090_alcrls_text[] = + { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; + +static const struct soc_enum max98090_alcrls_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_TIMING, M98090_ALCRLS_SHIFT, + ARRAY_SIZE(max98090_alcrls_text), max98090_alcrls_text); + +static const char * max98090_alccmp_text[] = + { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; + +static const struct soc_enum max98090_alccmp_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_CMPR, M98090_ALCCMP_SHIFT, + ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); + +static const char * max98090_alcexp_text[] = { "1:1", "2:1", "3:1" }; + +static const struct soc_enum max98090_alcexp_enum = + SOC_ENUM_SINGLE(M98090_REG_ALC_EXP, M98090_ALCEXP_SHIFT, + ARRAY_SIZE(max98090_alcexp_text), max98090_alcexp_text); + +static const char * max98090_perf_pwr_text[] = { "High Performance", "Low Power" }; +static const char * max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; + +static const struct soc_enum max98090_dac_perfmode_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CFG, M98090_DAC_PERFMODE_SHIFT, + ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); + +static const struct soc_enum max98090_dachp_enum = + SOC_ENUM_SINGLE(M98090_REG_DAC_CFG, M98090_DACHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_adchp_enum = + SOC_ENUM_SINGLE(M98090_REG_ADC_CFG, M98090_ADCHP_SHIFT, + ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); + +static const struct soc_enum max98090_zdenn_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LVL, M98090_ZDENN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct soc_enum max98090_vs2enn_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LVL, M98090_VS2ENN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct soc_enum max98090_vsenn_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LVL, M98090_VSENN_SHIFT, + ARRAY_SIZE(max98090_enableddisabled_inv_text), + max98090_enableddisabled_inv_text); + +static const struct snd_kcontrol_new max98090_snd_controls[] = { + SOC_SINGLE("MIC Bias VCM Bandgap", M98090_REG_BIAS_CNTL, + M98090_VCM_MODE_SHIFT, M98090_VCM_MODE_NUM - 1, 0), + + SOC_SINGLE("DMIC MIC Comp Filter Config", M98090_REG_MIC_CFG2, + M98090_DMIC_COMP_SHIFT, M98090_DMIC_COMP_NUM - 1, 0), + + SOC_ENUM_EXT("External MIC Mux", max98090_extmic_mux_enum, + max98090_extmic_mux_get, max98090_extmic_mux_set), + + SOC_SINGLE_EXT_TLV("MIC1 Boost Volume", + M98090_REG_LVL_MIC1, M98090_MIC_PA1EN_SHIFT, + M98090_MIC_PA1EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_EXT_TLV("MIC2 Boost Volume", + M98090_REG_LVL_MIC2, M98090_MIC_PA2EN_SHIFT, + M98090_MIC_PA2EN_NUM - 1, 0, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + + SOC_SINGLE_TLV("MIC1 Volume", M98090_REG_LVL_MIC1, + M98090_MIC_PGAM1_SHIFT, M98090_MIC_PGAM1_NUM - 1, 1, + max98090_mic_tlv), + + SOC_SINGLE_TLV("MIC2 Volume", M98090_REG_LVL_MIC2, + M98090_MIC_PGAM2_SHIFT, M98090_MIC_PGAM2_NUM - 1, 1, + max98090_mic_tlv), + + SOC_ENUM("LINEA Single Ended Reduction", max98090_mixg135_enum), + SOC_ENUM("LINEB Single Ended Reduction", max98090_mixg246_enum), + + SOC_SINGLE_RANGE_TLV("LINEA Volume", M98090_REG_LVL_LINE, + M98090_LINAPGA_SHIFT, 0, M98090_LINAPGA_NUM - 1, 1, + max98090_lin_tlv), + + SOC_SINGLE_RANGE_TLV("LINEB Volume", M98090_REG_LVL_LINE, + M98090_LINBPGA_SHIFT, 0, M98090_LINBPGA_NUM - 1, 1, + max98090_lin_tlv), + + SOC_SINGLE("LINEA Ext Resistor Gain Mode", M98090_REG_CFG_LINE, + M98090_EXTBUFA_SHIFT, M98090_EXTBUFA_NUM - 1, 0), + SOC_SINGLE("LINEB Ext Resistor Gain Mode", M98090_REG_CFG_LINE, + M98090_EXTBUFB_SHIFT, M98090_EXTBUFB_NUM - 1, 0), + + SOC_SINGLE_TLV("ADCL Boost Volume", M98090_REG_LVL_ADC_L, + M98090_ADC_AVLG_SHIFT, M98090_ADC_AVLG_NUM - 1, 0, + max98090_adcboost_tlv), + SOC_SINGLE_TLV("ADCR Boost Volume", M98090_REG_LVL_ADC_R, + M98090_ADC_AVRG_SHIFT, M98090_ADC_AVLG_NUM - 1, 0, + max98090_adcboost_tlv), + + SOC_SINGLE_TLV("ADCL Volume", M98090_REG_LVL_ADC_L, + M98090_ADC_AVL_SHIFT, M98090_ADC_AVL_NUM - 1, 1, + max98090_adc_tlv), + SOC_SINGLE_TLV("ADCR Volume", M98090_REG_LVL_ADC_R, + M98090_ADC_AVR_SHIFT, M98090_ADC_AVR_NUM - 1, 1, + max98090_adc_tlv), + + SOC_ENUM("ADC Oversampling Rate", max98090_osr128_enum), + SOC_SINGLE("ADC Quantizer Dither", M98090_REG_ADC_CFG, + M98090_ADCDITHER_SHIFT, M98090_ADCDITHER_NUM - 1, 0), + SOC_ENUM("ADC High Performance Mode", max98090_adchp_enum), + + SOC_SINGLE("DAC Mono Mode", M98090_REG_DAI_IOCFG, + M98090_DAI_DMONO_SHIFT, M98090_DAI_DMONO_NUM - 1, 0), + SOC_ENUM("SDIN Mode", max98090_sdien_enum), + SOC_ENUM("SDOUT Mode", max98090_sdoen_enum), + SOC_ENUM("SDOUT Hi-Z Mode", max98090_hizoff_enum), + SOC_ENUM("Filter Mode", max98090_filter_mode_enum), + SOC_SINGLE("Record Path DC Blocking", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_AHPF_SHIFT, M98090_DAI_FLT_AHPF_NUM - 1, 0), + SOC_SINGLE("Playback Path DC Blocking", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DHPF_SHIFT, M98090_DAI_FLT_DHPF_NUM - 1, 0), + SOC_SINGLE_TLV("Digital BQ Volume", M98090_REG_LVL_BIQUAD, + M98090_AVBQ_SHIFT, M98090_AVBQ_NUM - 1, 1, max98090_dv_tlv), + SOC_ENUM("Digital Sidetone Source", max98090_dsts_enum), + SOC_SINGLE_EXT_TLV("Digital Sidetone Volume", + M98090_REG_LVL_SIDETONE, M98090_DVST_SHIFT, + M98090_DVST_NUM - 1, 1, max98090_get_enab_tlv, + max98090_put_enab_tlv, max98090_micboost_tlv), + SOC_SINGLE_TLV("Digital Coarse Volume", M98090_REG_DAI_LVL, + M98090_DAI_LVL_DVG_SHIFT, M98090_DAI_LVL_DVG_NUM - 1, 0, + max98090_dvg_tlv), + SOC_SINGLE_TLV("Digital Volume", M98090_REG_DAI_LVL, + M98090_DAI_LVL_DV_SHIFT, M98090_DAI_LVL_DV_NUM - 1, 1, + max98090_dv_tlv), + SND_SOC_BYTES("EQ Coefficients", M98090_REG_EQ_BASE, 105), + SOC_SINGLE("Digital EQ 3 Band Switch", M98090_REG_DSP_EQ_EN, + M98090_EQ3BANDEN_SHIFT, M98090_EQ3BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 5 Band Switch", M98090_REG_DSP_EQ_EN, + M98090_EQ5BANDEN_SHIFT, M98090_EQ5BANDEN_NUM - 1, 0), + SOC_SINGLE("Digital EQ 7 Band Switch", M98090_REG_DSP_EQ_EN, + M98090_EQ7BANDEN_SHIFT, M98090_EQ7BANDEN_NUM - 1, 0), + SOC_ENUM("Digital EQ Clipping Detection", max98090_eqclpn_enum), + SOC_SINGLE_TLV("Digital EQ Volume", M98090_REG_DAI_LVL_EQ, + M98090_DAI_LVL_DVEQ_SHIFT, M98090_DAI_LVL_DVEQ_NUM - 1, 1, + max98090_dv_tlv), + + SOC_SINGLE("ALC Enable", M98090_REG_ALC_TIMING, + M98090_ALCEN_SHIFT, M98090_ALCEN_NUM - 1, 0), + SOC_ENUM("ALC Attack Time", max98090_alcatk_enum), + SOC_ENUM("ALC Release Time", max98090_alcrls_enum), + SOC_SINGLE_TLV("ALC Make Up Volume", M98090_REG_LVL_ALC, + M98090_ALCG_SHIFT, M98090_ALCG_NUM - 1, 0, + max98090_alcmakeup_tlv), + SOC_ENUM("ALC Compression Ratio", max98090_alccmp_enum), + SOC_ENUM("ALC Expansion Ratio", max98090_alcexp_enum), + SOC_SINGLE_TLV("ALC Compression Threshold Volume", + M98090_REG_ALC_CMPR, M98090_ALCTHC_SHIFT, + M98090_ALCTHC_NUM - 1, 1, max98090_alccomp_tlv), + SOC_SINGLE_TLV("ALC Expansion Threshold Volume", + M98090_REG_ALC_EXP, M98090_ALCTHE_SHIFT, + M98090_ALCTHE_NUM - 1, 1, max98090_alcexp_tlv), + + SOC_ENUM("DAC HP Playback Performance Mode", + max98090_dac_perfmode_enum), + SOC_ENUM("DAC High Performance Mode", max98090_dachp_enum), + + SOC_SINGLE_TLV("Headphone Left Mixer Volume", + M98090_REG_MIX_HP_CNTL, M98090_MIXHPLG_SHIFT, + M98090_MIXHPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Headphone Right Mixer Volume", + M98090_REG_MIX_HP_CNTL, M98090_MIXHPRG_SHIFT, + M98090_MIXHPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Speaker Left Mixer Volume", + M98090_REG_MIX_SPK_CNTL, M98090_MIXSPLG_SHIFT, + M98090_MIXSPLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Speaker Right Mixer Volume", + M98090_REG_MIX_SPK_CNTL, M98090_MIXSPRG_SHIFT, + M98090_MIXSPRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_SINGLE_TLV("Receiver Left Mixer Volume", + M98090_REG_MIX_RCV_CTRL_LEFT, M98090_MIXRCVLG_SHIFT, + M98090_MIXRCVLG_NUM - 1, 1, max98090_mixout_tlv), + SOC_SINGLE_TLV("Receiver Right Mixer Volume", + M98090_REG_MIX_RCV_CNTL_RIGHT, M98090_MIXRCVRG_SHIFT, + M98090_MIXRCVRG_NUM - 1, 1, max98090_mixout_tlv), + + SOC_DOUBLE_R_TLV("Headphone Volume", M98090_REG_LVL_HP_LEFT, + M98090_REG_LVL_HP_RIGHT, M98090_HPVOLL_SHIFT, + M98090_HPVOLL_NUM - 1, 0, max98090_hp_tlv), + + SOC_DOUBLE_R_RANGE_TLV("Speaker Volume", + M98090_REG_LVL_SPK_LEFT, M98090_REG_LVL_SPK_RIGHT, + M98090_SPVOLL_SHIFT, 24, M98090_SPVOLL_NUM - 1 + 24, + 0, max98090_spk_tlv), + + SOC_DOUBLE_R_TLV("Receiver Volume", M98090_REG_LVL_RCV_LEFT, + M98090_REG_LVL_RCV_RIGHT, M98090_RCVLVOL_SHIFT, + M98090_RCVLVOL_NUM - 1, 0, max98090_rcv_lout_tlv), + + SOC_SINGLE("Headphone Left Switch", M98090_REG_LVL_HP_LEFT, + M98090_HPLM_SHIFT, 1, 1), + SOC_SINGLE("Headphone Right Switch", M98090_REG_LVL_HP_RIGHT, + M98090_HPRM_SHIFT, 1, 1), + + SOC_SINGLE("Speaker Left Switch", M98090_REG_LVL_SPK_LEFT, + M98090_SPLM_SHIFT, 1, 1), + SOC_SINGLE("Speaker Right Switch", M98090_REG_LVL_SPK_RIGHT, + M98090_SPRM_SHIFT, 1, 1), + + SOC_SINGLE("Receiver Left Switch", M98090_REG_LVL_RCV_LEFT, + M98090_RCVLM_SHIFT, 1, 1), + SOC_SINGLE("Receiver Right Switch", M98090_REG_LVL_RCV_RIGHT, + M98090_RCVRM_SHIFT, 1, 1), + + SOC_ENUM("Zero-Crossing Detection", max98090_zdenn_enum), + SOC_ENUM("Enhanced Volume Smoothing", max98090_vs2enn_enum), + SOC_ENUM("Volume Adjustment Smoothing", max98090_vsenn_enum), + + SOC_SINGLE("Quick Setup System Clock", M98090_REG_QCFG_SYS_CLK, + M98090_CLK_ALL_SHIFT, M98090_CLK_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Sample Rate", M98090_REG_QCFG_RATE, + M98090_SR_ALL_SHIFT, M98090_SR_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup DAI Interface", M98090_REG_QCFG_DAI, + M98090_DAI_ALL_SHIFT, M98090_DAI_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup DAC Path", M98090_REG_QCFG_DAC, + M98090_DIG2_ALL_SHIFT, M98090_DIG2_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup MIC/Direct ADC", M98090_REG_QCFG_MIC_PATH, + M98090_MIC_ALL_SHIFT, M98090_MIC_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Line ADC", M98090_REG_QCFG_LINE_PATH, + M98090_LINE_ALL_SHIFT, M98090_LINE_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Analog MIC Loop", M98090_REG_QCFG_MIC_LOOP, + M98090_AMIC_ALL_SHIFT, M98090_AMIC_ALL_NUM - 1, 0), + SOC_SINGLE("Quick Setup Analog Line Loop", + M98090_REG_QCFG_LINE_LOOP, M98090_ALIN_ALL_SHIFT, + M98090_ALIN_ALL_NUM - 1, 0), + + SND_SOC_BYTES("Biquad Coefficients", M98090_REG_BIQUAD_BASE, 15), + SOC_SINGLE("Biquad Switch", M98090_REG_DSP_EQ_EN, + M98090_ADCBQEN_SHIFT, M98090_ADCBQEN_NUM - 1, 0), +}; + +static const struct snd_kcontrol_new max98091_snd_controls[] = { + + SOC_SINGLE("DMIC34 Zeropad", M98090_REG_SAMPLE_RATE, + M98090_DMIC34_ZEROPAD_SHIFT, + M98090_DMIC34_ZEROPAD_NUM - 1, 0), + + SOC_ENUM("Filter DMIC34 Mode", max98090_filter_dmic34mode_enum), + SOC_SINGLE("DMIC34 DC Blocking", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DMIC34HPF_SHIFT, + M98090_DAI_FLT_DMIC34HPF_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC3 Boost Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3G_SHIFT, M98090_DMIC_AV3G_NUM - 1, 0, + max98090_adcboost_tlv), + SOC_SINGLE_TLV("DMIC4 Boost Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4G_SHIFT, M98090_DMIC_AV4G_NUM - 1, 0, + max98090_adcboost_tlv), + + SOC_SINGLE_TLV("DMIC3 Volume", M98090_REG_DMIC3_VOLUME, + M98090_DMIC_AV3_SHIFT, M98090_DMIC_AV3_NUM - 1, 1, + max98090_adc_tlv), + SOC_SINGLE_TLV("DMIC4 Volume", M98090_REG_DMIC4_VOLUME, + M98090_DMIC_AV4_SHIFT, M98090_DMIC_AV4_NUM - 1, 1, + max98090_adc_tlv), + + SND_SOC_BYTES("DMIC34 Biquad Coefficients", + M98090_REG_DMIC34_BIQUAD_BASE, 15), + SOC_SINGLE("DMIC34 Biquad Switch", M98090_REG_DSP_EQ_EN, + M98090_DMIC34BQEN_SHIFT, M98090_DMIC34BQEN_NUM - 1, 0), + + SOC_SINGLE_TLV("DMIC34 BQ PreAttenuation Volume", + M98090_REG_DMIC34_BQ_PREATTEN, M98090_AV34BQ_SHIFT, + M98090_AV34BQ_NUM - 1, 1, max98090_dv_tlv), + + SOC_SINGLE("Rev ID", M98090_REG_REV_ID, + M98090_REVID_SHIFT, M98090_REVID_NUM - 1, 0), +}; + +static int max98090_mic_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event, + int channel) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, M98090_REG_CFG_LINE); + + val = (val & M98090_EXTMIC_MASK) >> M98090_EXTMIC_SHIFT; + + if ((val == (M98090_EXTMIC_MIC1 >> M98090_EXTMIC_SHIFT)) || + (val == (M98090_EXTMIC_MIC2 >> M98090_EXTMIC_SHIFT))) + { + max98090->extmic_mux = val - 1; /* Update for volatile */ + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to selected channel */ + max98090->extmic_mux = channel; + val = max98090->extmic_mux + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = M98090_EXTMIC_NONE >> M98090_EXTMIC_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, M98090_REG_CFG_LINE, M98090_EXTMIC_MASK, + val << M98090_EXTMIC_SHIFT); + + return 0; +} + +static int max98090_mic1_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return max98090_mic_mux_event(w, kcontrol, event, 0); +} + +static int max98090_mic2_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return max98090_mic_mux_event(w, kcontrol, event, 1); +} + +static int max98090_micinput_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + unsigned int val = snd_soc_read(codec, w->reg); + + if (w->reg == M98090_REG_LVL_MIC1) + val = (val & M98090_MIC_PA1EN_MASK) >> M98090_MIC_PA1EN_SHIFT; + else + val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; + + + if (val >= 1) { + if (w->reg == M98090_REG_LVL_MIC1) { + max98090->mic1pre = val - 1; /* Update for volatile */ + } else { + max98090->mic2pre = val - 1; /* Update for volatile */ + } + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* If turning on, set to most recently selected volume */ + if (w->reg == M98090_REG_LVL_MIC1) + val = max98090->mic1pre + 1; + else + val = max98090->mic2pre + 1; + break; + case SND_SOC_DAPM_POST_PMD: + /* If turning off, turn off */ + val = 0; + break; + default: + return -EINVAL; + } + + if (w->reg == M98090_REG_LVL_MIC1) + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA1EN_MASK, + val << M98090_MIC_PA1EN_SHIFT); + else + snd_soc_update_bits(codec, w->reg, M98090_MIC_PA2EN_MASK, + val << M98090_MIC_PA2EN_SHIFT); + + return 0; +} + +static int max98090_sidetone_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, w->reg, M98090_DVST_MASK, + (1 + max98090->sidetone) << M98090_DVST_SHIFT); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, M98090_DVST_MASK, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char *mic1_mux_text[] = { "IN12", "IN56" }; + +static const struct soc_enum mic1_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC1_SHIFT, + ARRAY_SIZE(mic1_mux_text), mic1_mux_text); + +static const struct snd_kcontrol_new max98090_mic1_mux = + SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); + +static const char *mic2_mux_text[] = { "IN34", "IN56" }; + +static const struct soc_enum mic2_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC2_SHIFT, + ARRAY_SIZE(mic2_mux_text), mic2_mux_text); + +static const struct snd_kcontrol_new max98090_mic2_mux = + SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); + +static const char * max98090_micpre_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_mic1pre_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_MIC1, M98090_MIC_PA1EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const struct soc_enum max98090_mic2pre_enum = + SOC_ENUM_SINGLE(M98090_REG_LVL_MIC2, M98090_MIC_PA2EN_SHIFT, + ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); + +static const char * max98090_extmic_text[] = { "Off", "On" }; + +static const struct soc_enum max98090_extmic_enum = + SOC_ENUM_SINGLE(M98090_REG_CFG_LINE, M98090_EXTMIC_SHIFT, + ARRAY_SIZE(max98090_extmic_text), max98090_extmic_text); + +static const struct snd_kcontrol_new max98090_extmic = + SOC_DAPM_ENUM("External MIC", max98090_extmic_enum); + +/* LINEA mixer switch */ +static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1 Switch", M98090_REG_CFG_INPUT, + M98090_IN1SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN3 Switch", M98090_REG_CFG_INPUT, + M98090_IN3SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN5 Switch", M98090_REG_CFG_INPUT, + M98090_IN5SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_CFG_INPUT, + M98090_IN34DIFF_SHIFT, 1, 0), +}; + +/* LINEB mixer switch */ +static const struct snd_kcontrol_new max98090_lineb_mixer_controls[] = { + SOC_DAPM_SINGLE("IN2 Switch", M98090_REG_CFG_INPUT, + M98090_IN2SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN4 Switch", M98090_REG_CFG_INPUT, + M98090_IN4SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN6 Switch", M98090_REG_CFG_INPUT, + M98090_IN6SEEN_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_CFG_INPUT, + M98090_IN56DIFF_SHIFT, 1, 0), +}; + +/* Left ADC mixer switch */ +static const struct snd_kcontrol_new max98090_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_ADC_L, + M98090_MIXADL_MIC2_SHIFT, 1, 0), +}; + +/* Right ADC mixer switch */ +static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("IN12 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_IN12DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN34 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_IN34DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("IN56 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_IN65DIFF_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_ADC_R, + M98090_MIXADR_MIC2_SHIFT, 1, 0), +}; + +static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; + +static const struct soc_enum ltenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct soc_enum ltenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LTEN_SHIFT, + ARRAY_SIZE(lten_mux_text), lten_mux_text); + +static const struct snd_kcontrol_new max98090_ltenl_mux = + SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); + +static const struct snd_kcontrol_new max98090_ltenr_mux = + SOC_DAPM_ENUM("LTENR Mux", ltenr_mux_enum); + +static const char *lben_mux_text[] = { "Normal", "Loopback" }; + +static const struct soc_enum lbenl_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct soc_enum lbenr_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_DAI_IOCFG, M98090_DAI_LBEN_SHIFT, + ARRAY_SIZE(lben_mux_text), lben_mux_text); + +static const struct snd_kcontrol_new max98090_lbenl_mux = + SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); + +static const struct snd_kcontrol_new max98090_lbenr_mux = + SOC_DAPM_ENUM("LBENR Mux", lbenr_mux_enum); + +/* Left speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_left_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_SPK_LEFT, + M98090_MIXSPL_MIC2_SHIFT, 1, 0), +}; + +/* Right speaker mixer switch */ +static const struct + snd_kcontrol_new max98090_right_speaker_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_SPK_RIGHT, + M98090_MIXSPR_MIC2_SHIFT, 1, 0), +}; + +/* Left headphone mixer switch */ +static const struct snd_kcontrol_new max98090_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_HP_LEFT, + M98090_MIXHPL_MIC2_SHIFT, 1, 0), +}; + +/* Right headphone mixer switch */ +static const struct snd_kcontrol_new max98090_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_HP_RIGHT, + M98090_MIXHPR_MIC2_SHIFT, 1, 0), +}; + +/* Left receiver mixer switch */ +static const struct snd_kcontrol_new max98090_left_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_RCV_LEFT, + M98090_MIXRCVL_MIC2_SHIFT, 1, 0), +}; + +/* Right receiver mixer switch */ +static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_DACL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("Right DAC Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_DACR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEA Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_LINEA_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LINEB Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_LINEB_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC1 Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_MIC1_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MIC2 Switch", M98090_REG_MIX_RCV_RIGHT, + M98090_MIXRCVR_MIC2_SHIFT, 1, 0), +}; + +static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; + +static const struct soc_enum linmod_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_MIX_RCV_RIGHT, M98090_LINMOD_SHIFT, + ARRAY_SIZE(linmod_mux_text), linmod_mux_text); + +static const struct snd_kcontrol_new max98090_linmod_mux = + SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); + +static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; + +/* + * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable + */ +static const struct soc_enum mixhplsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_MIX_HP_CNTL, M98090_MIXHPLSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhplsel_mux = + SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); + +static const struct soc_enum mixhprsel_mux_enum = + SOC_ENUM_SINGLE(M98090_REG_MIX_HP_CNTL, M98090_MIXHPRSEL_SHIFT, + ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); + +static const struct snd_kcontrol_new max98090_mixhprsel_mux = + SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); + +static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("DMICL"), + SND_SOC_DAPM_INPUT("DMICR"), + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN12"), + SND_SOC_DAPM_INPUT("IN34"), + SND_SOC_DAPM_INPUT("IN56"), + + SND_SOC_DAPM_MICBIAS("MICBIAS", M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_SHIFT, 0), + SND_SOC_DAPM_SUPPLY("SHDN", M98090_REG_PWR_SYS, + M98090_PWR_SHDNN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("VCM", M98090_REG_BIAS_CNTL, + M98090_VCM_MODE_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDIEN", M98090_REG_DAI_IOCFG, + M98090_DAI_SDIEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_DAI_IOCFG, + M98090_DAI_SDOEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMICL_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMICR_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_AHPF_SHIFT, 0, NULL, 0), + +/* + * Note: Sysclk and misc power supplies are taken care of by SHDN and VCM + */ + + SND_SOC_DAPM_MUX_E("MIC1 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic1_mux, + max98090_mic1_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MUX_E("MIC2 Mux", SND_SOC_NOPM, + 0, 0, &max98090_mic2_mux, + max98090_mic2_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC1 Input", M98090_REG_LVL_MIC1, + M98090_MIC_PA1EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA_E("MIC2 Input", M98090_REG_LVL_MIC2, + M98090_MIC_PA2EN_SHIFT, 0, NULL, 0, max98090_micinput_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_MIXER("LINEA Mixer", SND_SOC_NOPM, 0, 0, + &max98090_linea_mixer_controls[0], + ARRAY_SIZE(max98090_linea_mixer_controls)), + + SND_SOC_DAPM_MIXER("LINEB Mixer", SND_SOC_NOPM, 0, 0, + &max98090_lineb_mixer_controls[0], + ARRAY_SIZE(max98090_lineb_mixer_controls)), + + SND_SOC_DAPM_PGA("LINEA Input", M98090_REG_PWR_EN_IN, + M98090_PWR_LINEAEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("LINEB Input", M98090_REG_PWR_EN_IN, + M98090_PWR_LINEBEN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_adc_mixer_controls[0], + ARRAY_SIZE(max98090_left_adc_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_adc_mixer_controls[0], + ARRAY_SIZE(max98090_right_adc_mixer_controls)), + + SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_PWR_EN_IN, + M98090_PWR_ADLEN_SHIFT, 0), + SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_PWR_EN_IN, + M98090_PWR_ADREN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFOUTR", "HiFi Capture", 1, + SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("LBENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenl_mux), + + SND_SOC_DAPM_MUX("LBENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_lbenr_mux), + + SND_SOC_DAPM_MUX("LTENL Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenl_mux), + + SND_SOC_DAPM_MUX("LTENR Mux", SND_SOC_NOPM, + 0, 0, &max98090_ltenr_mux), + + SND_SOC_DAPM_PGA_E("Sidetone", M98090_REG_LVL_SIDETONE, + M98090_DVST_SHIFT, 0, NULL, 0, max98090_sidetone_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_AIF_IN("AIFINL", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFINR", "HiFi Playback", 1, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DACL", NULL, M98090_REG_PWR_EN_OUT, + M98090_PWR_DALEN_SHIFT, 0), + SND_SOC_DAPM_DAC("DACR", NULL, M98090_REG_PWR_EN_OUT, + M98090_PWR_DAREN_SHIFT, 0), + + SND_SOC_DAPM_MIXER("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_hp_mixer_controls[0], + ARRAY_SIZE(max98090_left_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_hp_mixer_controls[0], + ARRAY_SIZE(max98090_right_hp_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_left_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Speaker Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_speaker_mixer_controls[0], + ARRAY_SIZE(max98090_right_speaker_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_left_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_left_rcv_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Receiver Mixer", SND_SOC_NOPM, 0, 0, + &max98090_right_rcv_mixer_controls[0], + ARRAY_SIZE(max98090_right_rcv_mixer_controls)), + + SND_SOC_DAPM_MUX("LINMOD Mux", M98090_REG_MIX_RCV_RIGHT, + M98090_LINMOD_SHIFT, 0, &max98090_linmod_mux), + + SND_SOC_DAPM_MUX("MIXHPLSEL Mux", M98090_REG_MIX_HP_CNTL, + M98090_MIXHPLSEL_SHIFT, 0, &max98090_mixhplsel_mux), + + SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_MIX_HP_CNTL, + M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), + + SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_HPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_HPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_SPLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK Right Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_SPREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_PGA("RCV Left Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_RCVLEN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("RCV Right Out", M98090_REG_PWR_EN_OUT, + M98090_PWR_RCVREN_SHIFT, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("RCVL"), + SND_SOC_DAPM_OUTPUT("RCVR"), +}; + +static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + + SND_SOC_DAPM_SUPPLY("DMIC3_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMIC3_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC4_ENA", M98090_REG_MIC_CFG1, + M98090_DIGMIC4_SHIFT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route max98090_dapm_routes[] = { + + {"MIC1 Input", NULL, "MIC1"}, + {"MIC2 Input", NULL, "MIC2"}, + {"MIC1 Input", NULL, "MICBIAS"}, + {"MIC2 Input", NULL, "MICBIAS"}, + + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, + {"DMICL", NULL, "AHPF"}, + {"DMICR", NULL, "AHPF"}, + + /* MIC1 input mux */ + {"MIC1 Mux", "IN12", "IN12"}, + {"MIC1 Mux", "IN56", "IN56"}, + + /* MIC2 input mux */ + {"MIC2 Mux", "IN34", "IN34"}, + {"MIC2 Mux", "IN56", "IN56"}, + + {"MIC1 Input", NULL, "MIC1 Mux"}, + {"MIC2 Input", NULL, "MIC2 Mux"}, + + /* Left ADC input mixer */ + {"Left ADC Mixer", "IN12 Switch", "IN12"}, + {"Left ADC Mixer", "IN34 Switch", "IN34"}, + {"Left ADC Mixer", "IN56 Switch", "IN56"}, + {"Left ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Left ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Left ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Right ADC input mixer */ + {"Right ADC Mixer", "IN12 Switch", "IN12"}, + {"Right ADC Mixer", "IN34 Switch", "IN34"}, + {"Right ADC Mixer", "IN56 Switch", "IN56"}, + {"Right ADC Mixer", "LINEA Switch", "LINEA Input"}, + {"Right ADC Mixer", "LINEB Switch", "LINEB Input"}, + {"Right ADC Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right ADC Mixer", "MIC2 Switch", "MIC2 Input"}, + + /* Line A input mixer */ + {"LINEA Mixer", "IN1 Switch", "IN1"}, + {"LINEA Mixer", "IN3 Switch", "IN3"}, + {"LINEA Mixer", "IN5 Switch", "IN5"}, + {"LINEA Mixer", "IN34 Switch", "IN34"}, + + /* Line B input mixer */ + {"LINEB Mixer", "IN2 Switch", "IN2"}, + {"LINEB Mixer", "IN4 Switch", "IN4"}, + {"LINEB Mixer", "IN6 Switch", "IN6"}, + {"LINEB Mixer", "IN56 Switch", "IN56"}, + + {"LINEA Input", NULL, "LINEA Mixer"}, + {"LINEB Input", NULL, "LINEB Mixer"}, + + /* Inputs */ + {"ADCL", NULL, "Left ADC Mixer"}, + {"ADCR", NULL, "Right ADC Mixer"}, + {"ADCL", NULL, "SHDN"}, + {"ADCR", NULL, "SHDN"}, + {"ADCL", NULL, "VCM"}, + {"ADCR", NULL, "VCM"}, + + {"LBENL Mux", "Normal", "ADCL"}, + {"LBENL Mux", "Normal", "DMICL"}, + {"LBENL Mux", "Loopback", "LTENL Mux"}, + {"LBENR Mux", "Normal", "ADCR"}, + {"LBENR Mux", "Normal", "DMICR"}, + {"LBENR Mux", "Loopback", "LTENR Mux"}, + + {"AIFOUTL", NULL, "LBENL Mux"}, + {"AIFOUTR", NULL, "LBENR Mux"}, + {"AIFOUTL", NULL, "SHDN"}, + {"AIFOUTR", NULL, "SHDN"}, + {"AIFOUTL", NULL, "VCM"}, + {"AIFOUTR", NULL, "VCM"}, + {"AIFOUTL", NULL, "SDOEN"}, + {"AIFOUTR", NULL, "SDOEN"}, + + {"LTENL Mux", "Normal", "AIFINL"}, + {"LTENL Mux", "Loopthrough", "LBENL Mux"}, + {"LTENR Mux", "Normal", "AIFINR"}, + {"LTENR Mux", "Loopthrough", "LBENR Mux"}, + + {"DACL", NULL, "LTENL Mux"}, + {"DACR", NULL, "LTENR Mux"}, + + {"AIFINL", NULL, "SHDN"}, + {"AIFINR", NULL, "SHDN"}, + {"AIFINL", NULL, "VCM"}, + {"AIFINR", NULL, "VCM"}, + {"AIFINL", NULL, "SDIEN"}, + {"AIFINR", NULL, "SDIEN"}, + {"DACL", NULL, "SHDN"}, + {"DACR", NULL, "SHDN"}, + {"DACL", NULL, "VCM"}, + {"DACR", NULL, "VCM"}, + + /* Left headphone output mixer */ + {"Left Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Left Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Left Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right headphone output mixer */ + {"Right Headphone Mixer", "Left DAC Switch", "DACL"}, + {"Right Headphone Mixer", "Right DAC Switch", "DACR"}, + {"Right Headphone Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Headphone Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Headphone Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Headphone Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left speaker output mixer */ + {"Left Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Left Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Left Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right speaker output mixer */ + {"Right Speaker Mixer", "Left DAC Switch", "DACL"}, + {"Right Speaker Mixer", "Right DAC Switch", "DACR"}, + {"Right Speaker Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Speaker Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Speaker Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Speaker Mixer", "LINEB Switch", "LINEB Input"}, + + /* Left Receiver output mixer */ + {"Left Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Left Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Left Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Left Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Left Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Left Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + /* Right Receiver output mixer */ + {"Right Receiver Mixer", "Left DAC Switch", "DACL"}, + {"Right Receiver Mixer", "Right DAC Switch", "DACR"}, + {"Right Receiver Mixer", "MIC1 Switch", "MIC1 Input"}, + {"Right Receiver Mixer", "MIC2 Switch", "MIC2 Input"}, + {"Right Receiver Mixer", "LINEA Switch", "LINEA Input"}, + {"Right Receiver Mixer", "LINEB Switch", "LINEB Input"}, + + {"MIXHPLSEL Mux", "HP Mixer", "Left Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Left Out", NULL, "DACL"}, + {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + + {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, + + /* + * Disable this for lowest power if bypassing + * the DAC with an analog signal + */ + {"HP Right Out", NULL, "DACR"}, + {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + + {"SPK Left Out", NULL, "Left Speaker Mixer"}, + {"SPK Right Out", NULL, "Right Speaker Mixer"}, + {"RCV Left Out", NULL, "Left Receiver Mixer"}, + + {"LINMOD Mux", "Left and Right", "Right Receiver Mixer"}, + {"LINMOD Mux", "Left Only", "Left Receiver Mixer"}, + {"RCV Right Out", NULL, "LINMOD Mux"}, + + {"HPL", NULL, "HP Left Out"}, + {"HPR", NULL, "HP Right Out"}, + {"SPKL", NULL, "SPK Left Out"}, + {"SPKR", NULL, "SPK Right Out"}, + {"RCVL", NULL, "RCV Left Out"}, + {"RCVR", NULL, "RCV Right Out"}, + +}; + +static const struct snd_soc_dapm_route max98091_dapm_routes[] = { + + /* DMIC inputs */ + {"DMIC3", NULL, "DMIC3_ENA"}, + {"DMIC4", NULL, "DMIC4_ENA"}, + {"DMIC3", NULL, "AHPF"}, + {"DMIC4", NULL, "AHPF"}, + +}; + +static int max98090_add_widgets(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_add_codec_controls(codec, max98090_snd_controls, + ARRAY_SIZE(max98090_snd_controls)); + + if (max98090->devtype == MAX98091) { + snd_soc_add_codec_controls(codec, max98091_snd_controls, + ARRAY_SIZE(max98091_snd_controls)); + } + + snd_soc_dapm_new_controls(dapm, max98090_dapm_widgets, + ARRAY_SIZE(max98090_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98090_dapm_routes, + ARRAY_SIZE(max98090_dapm_routes)); + + if (max98090->devtype == MAX98091) { + snd_soc_dapm_new_controls(dapm, max98091_dapm_widgets, + ARRAY_SIZE(max98091_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, + ARRAY_SIZE(max98091_dapm_routes)); + + } + + return 0; +} + +static const int pclk_rates[] = { + 12000000, 12000000, 13000000, 13000000, + 16000000, 16000000, 19200000, 19200000 +}; + +static const int lrclk_rates[] = { + 8000, 16000, 8000, 16000, + 8000, 16000, 8000, 16000 +}; + +static const int user_pclk_rates[] = { + 13000000, 13000000 +}; + +static const int user_lrclk_rates[] = { + 44100, 48000 +}; + +static const unsigned long long ni_value[] = { + 3528, 768 +}; + +static const unsigned long long mi_value[] = { + 8125, 1625 +}; + +static void max98090_configure_bclk(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + unsigned long long ni; + int i; + + if (!max98090->sysclk) { + dev_err(codec->dev, "No SYSCLK configured\n"); + return; + } + + if (!max98090->bclk || !max98090->lrclk) { + dev_err(codec->dev, "No audio clocks configured\n"); + return; + } + + /* Skip configuration when operating as slave */ + if (!(snd_soc_read(codec, M98090_REG_CLOCK_MAS_MODE) & + M98090_CLK_MAS_MASK)) { + return; + } + + /* Check for supported PCLK to LRCLK ratios */ + for (i = 0; i < ARRAY_SIZE(pclk_rates); i++) { + if ((pclk_rates[i] == max98090->sysclk) && + (lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found supported PCLK to LRCLK rates 0x%x\n", + i + 0x8); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_FREQ1_MASK, + (i + 0x8) << M98090_CLK_FREQ1_SHIFT); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, 0); + return; + } + } + + /* Check for user calculated MI and NI ratios */ + for (i = 0; i < ARRAY_SIZE(user_pclk_rates); i++) { + if ((user_pclk_rates[i] == max98090->sysclk) && + (user_lrclk_rates[i] == max98090->lrclk)) { + dev_dbg(codec->dev, + "Found user supported PCLK to LRCLK rates\n"); + dev_dbg(codec->dev, "i %d ni %lld mi %lld\n", + i, ni_value[i], mi_value[i]); + + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_FREQ1_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, + 1 << M98090_CLK_USE_M1_SHIFT); + + snd_soc_write(codec, M98090_REG_CLOCK_DAI1_NI_HI, + (ni_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_DAI2_NI_LO, + ni_value[i] & 0xFF); + snd_soc_write(codec, M98090_REG_CLOCK_DAI3_MI_HI, + (mi_value[i] >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_DAI4_MI_LO, + mi_value[i] & 0xFF); + + return; + } + } + + /* + * Calculate based on MI = 65536 (not as good as either method above) + */ + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_FREQ1_MASK, 0); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, 0); + + /* + * Configure NI when operating as master + * Note: There is a small, but significant audio quality improvement + * by calculating ni and mi. + */ + ni = 65536ULL * (max98090->lrclk < 50000 ? 96ULL : 48ULL) + * (unsigned long long int)max98090->lrclk; + do_div(ni, (unsigned long long int)max98090->sysclk); + dev_info(codec->dev, "No better method found\n"); + dev_info(codec->dev, "Calculating ni %lld with mi 65536\n", ni); + snd_soc_write(codec, M98090_REG_CLOCK_DAI1_NI_HI, + (ni >> 8) & 0x7F); + snd_soc_write(codec, M98090_REG_CLOCK_DAI2_NI_LO, ni & 0xFF); +} + +static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + u8 regval; + + max98090->dai_fmt = fmt; + cdata = &max98090->dai[0]; + + if (fmt != cdata->fmt) { + cdata->fmt = fmt; + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Set to slave mode PLL - MAS mode off */ + snd_soc_write(codec, + M98090_REG_CLOCK_DAI1_NI_HI, 0x00); + snd_soc_write(codec, + M98090_REG_CLOCK_DAI2_NI_LO, 0x00); + snd_soc_update_bits(codec, M98090_REG_CLOCK_MODE, + M98090_CLK_USE_M1_MASK, 0); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* Set to master mode */ + if (max98090->tdm_slots == 4) { + /* TDM */ + regval |= M98090_CLK_MAS_MASK | + M98090_CLK_BSEL_64; + } else if (max98090->tdm_slots == 3) { + /* TDM */ + regval |= M98090_CLK_MAS_MASK | + M98090_CLK_BSEL_48; + } else { + /* Few TDM slots, or No TDM */ + regval |= M98090_CLK_MAS_MASK | + M98090_CLK_BSEL_32; + } + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + snd_soc_write(codec, M98090_REG_CLOCK_MAS_MODE, regval); + + regval = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regval |= M98090_DAI_DLY_MASK; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_RIGHT_J: + regval |= M98090_DAI_RJ_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Not supported mode */ + default: + dev_err(codec->dev, "DAI format unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + regval |= M98090_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + regval |= M98090_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + regval |= M98090_DAI_BCI_MASK|M98090_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + +/* + * This accommodates an inverted logic in the MAX98090 chip + * for Bit Clock Invert (BCI). The inverted logic is only seen + * for the case of TDM mode. The remaining cases have normal logic. + */ + if (max98090->tdm_slots > 1) { + regval ^= M98090_DAI_BCI_MASK; + } + + snd_soc_write(codec, + M98090_REG_DAI_INTERFACE_FORMAT, regval); + } + + return 0; +} + +static int max98090_set_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + cdata = &max98090->dai[0]; + + if (slots < 0 || slots > 4) + return -EINVAL; + + max98090->tdm_slots = slots; + max98090->tdm_width = slot_width; + + if (max98090->tdm_slots > 1) { + /* SLOTL SLOTR SLOTDLY */ + snd_soc_write(codec, M98090_REG_DAI_TDM_FORMAT, + 0 << M98090_DAI_TDM_SLOTL_SHIFT | + 1 << M98090_DAI_TDM_SLOTR_SHIFT | + 0 << M98090_DAI_TDM_SLOTDLY_SHIFT); + + /* FSW TDM */ + snd_soc_update_bits(codec, M98090_REG_DAI_TDM_CONTROL, + M98090_DAI_TDM_MASK, + M98090_DAI_TDM_MASK); + } + + /* + * Normally advisable to set TDM first, but this permits either order + */ + cdata->fmt = 0; + max98090_dai_set_fmt(codec_dai, max98090->dai_fmt); + + return 0; +} + +static int max98090_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { + /* + * Set to normal bias level. + */ + snd_soc_update_bits(codec, M98090_REG_MIC_BIAS, + M98090_MBVSEL_MASK, M98090_MBVSEL_2V4); + + snd_soc_update_bits(codec, M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_MASK, M98090_PWR_MBEN_MASK); + } + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_MASK, 0); + /* Set internal pull-up to lowest power mode */ + snd_soc_update_bits(codec, M98090_REG_CFG_JACK, + M98090_JDWK_MASK, M98090_JDWK_MASK); + regcache_mark_dirty(max98090->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const int comp_pclk_rates[] = { + 11289600, 12288000, 12000000, 13000000, 19200000 +}; + +static const int dmic_micclk[] = { + 2, 2, 2, 2, 4, 2 +}; + +static const int comp_lrclk_rates[] = { + 8000, 16000, 32000, 44100, 48000, 96000 +}; + +static const int dmic_comp[6][6] = { + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 3, 3, 3}, + {7, 8, 3, 1, 1, 1}, + {7, 8, 3, 1, 2, 2}, + {7, 8, 3, 3, 3, 3} +}; + +static int max98090_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int i, j; + + cdata = &max98090->dai[0]; + max98090->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + max98090->bclk *= 2; + + max98090->lrclk = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_update_bits(codec, M98090_REG_DAI_INTERFACE_FORMAT, + M98090_DAI_WS_MASK, 0); + break; + default: + return -EINVAL; + } + + max98090_configure_bclk(codec); + + cdata->rate = max98090->lrclk; + + /* Update filter mode */ + if (max98090->lrclk < 24000) + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_MODE_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_MODE_MASK, M98090_DAI_FLT_MODE_MASK); + + /* Update sample rate mode */ + if (max98090->lrclk < 50000) + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DHF_MASK, 0); + else + snd_soc_update_bits(codec, M98090_REG_DAI_FILTERS, + M98090_DAI_FLT_DHF_MASK, M98090_DAI_FLT_DHF_MASK); + + /* Check for supported PCLK to LRCLK ratios */ + for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) { + if (comp_pclk_rates[j] == max98090->sysclk) { + break; + } + } + + for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) { + if (max98090->lrclk <= (comp_lrclk_rates[i] + + comp_lrclk_rates[i + 1]) / 2) { + break; + } + } + + snd_soc_update_bits(codec, M98090_REG_MIC_CFG1, + M98090_MICCLK_MASK, + dmic_micclk[j] << M98090_MICCLK_SHIFT); + + snd_soc_update_bits(codec, M98090_REG_MIC_CFG2, + M98090_DMIC_COMP_MASK, + dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT); + + return 0; +} + +/* + * PLL / Sysclk + */ +static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + /* Requested clock frequency is already setup */ + if (freq == max98090->sysclk) + return 0; + + /* Setup clocks for slave mode, and using the PLL + * PSCLK = 0x01 (when master clk is 10MHz to 20MHz) + * 0x02 (when master clk is 20MHz to 40MHz).. + * 0x03 (when master clk is 40MHz to 60MHz).. + */ + if ((freq >= 10000000) && (freq < 20000000)) { + snd_soc_write(codec, M98090_REG_SYS_CLOCK, + M98090_CLK_PSCLK_DIV1); + } else if ((freq >= 20000000) && (freq < 40000000)) { + snd_soc_write(codec, M98090_REG_SYS_CLOCK, + M98090_CLK_PSCLK_DIV2); + } else if ((freq >= 40000000) && (freq < 60000000)) { + snd_soc_write(codec, M98090_REG_SYS_CLOCK, + M98090_CLK_PSCLK_DIV4); + } else { + dev_err(codec->dev, "Invalid master clock frequency\n"); + return -EINVAL; + } + + max98090->sysclk = freq; + + max98090_configure_bclk(codec); + + return 0; +} + +static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int regval; + + regval = mute ? M98090_DAI_LVL_DVM_MASK : 0; + snd_soc_update_bits(codec, M98090_REG_DAI_LVL, + M98090_DAI_LVL_DVM_MASK, regval); + + return 0; +} + +static void max98090_jack_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = container_of(work, + struct max98090_priv, + jack_work.work); + struct snd_soc_codec *codec = max98090->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int status = 0; + int reg; + + /* Read a second time */ + if (max98090->jack_state == M98090_JACK_STATE_NO_HEADSET) { + + snd_soc_update_bits(codec, M98090_REG_PWR_EN_IN, + M98090_PWR_MBEN_MASK, 0); + + /* Strong pull up allows mic detection */ + snd_soc_update_bits(codec, M98090_REG_CFG_JACK, + M98090_JDWK_MASK, 0); + + msleep(50); + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + /* Weak pull up allows only insertion detection */ + snd_soc_update_bits(codec, M98090_REG_CFG_JACK, + M98090_JDWK_MASK, M98090_JDWK_MASK); + } else { + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + } + + reg = snd_soc_read(codec, M98090_REG_JACK_STATUS); + + switch (reg & (M98090_LSNS_MASK | M98090_JKSNS_MASK)) { + case M98090_LSNS_MASK | M98090_JKSNS_MASK: + { + dev_info(codec->dev, "No Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + max98090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + status |= 0; + + break; + } + + case 0: + { + if (max98090->jack_state == + M98090_JACK_STATE_HEADSET) { + + dev_info(codec->dev, + "Headset Button Down Detected\n"); + + /* + * max98090_headset_button_event(codec) + * could be defined, then called here. + */ + + status |= SND_JACK_HEADSET; + status |= SND_JACK_BTN_0; + + break; + } + + /* Line is reported as Headphone */ + /* Nokia Headset is reported as Headphone */ + /* Mono Headphone is reported as Headphone */ + dev_info(codec->dev, "Headphone Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADPHONE; + + max98090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + status |= SND_JACK_HEADPHONE; + + break; + } + + case M98090_JKSNS_MASK: + { + dev_info(codec->dev, "Headset Detected\n"); + + max98090->jack_state = M98090_JACK_STATE_HEADSET; + + max98090_set_bias_level(codec, SND_SOC_BIAS_ON); + + status |= SND_JACK_HEADSET; + + break; + } + + default: + { + dev_info(codec->dev, "Unrecognized Jack Status\n"); + break; + } + } + + snd_soc_jack_report(max98090->jack, status, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + snd_soc_dapm_sync(dapm); +} + +static irqreturn_t max98090_interrupt(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + int ret; + unsigned int mask; + unsigned int active; + + dev_dbg(codec->dev, "***** max98090_interrupt *****\n"); + + ret = regmap_read(max98090->regmap, M98090_REG_IRQ_ENABLE, &mask); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_IRQ_ENABLE: %d\n", + ret); + return IRQ_NONE; + } + + ret = regmap_read(max98090->regmap, M98090_REG_IRQ_STATUS, &active); + + if (ret != 0) { + dev_err(codec->dev, + "failed to read M98090_REG_IRQ_STATUS: %d\n", + ret); + return IRQ_NONE; + } + + dev_dbg(codec->dev, "active=0x%02x mask=0x%02x -> active=0x%02x\n", + active, mask, active & mask); + + active &= mask; + + if (!active) + return IRQ_NONE; + + if (active & M98090_IRQ_CLD_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_CLD_MASK\n"); + } + + if (active & M98090_IRQ_SLD_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_SLD_MASK\n"); + } + + if (active & M98090_IRQ_ULK_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_ULK_MASK\n"); + } + + if (active & M98090_IRQ_JDET_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_JDET_MASK\n"); + + pm_wakeup_event(codec->dev, 100); + + schedule_delayed_work(&max98090->jack_work, + msecs_to_jiffies(100)); + } + + if (active & M98090_IRQ_ALCACT_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_ALCACT_MASK\n"); + } + + if (active & M98090_IRQ_ALCCLP_MASK) { + dev_dbg(codec->dev, "M98090_IRQ_ALCCLP_MASK\n"); + } + + return IRQ_HANDLED; +} + +/** + * max98090_mic_detect - Enable microphone detection via the MAX98090 IRQ + * + * @codec: MAX98090 codec + * @jack: jack to report detection events on + * + * Enable microphone detection via IRQ on the MAX98090. If GPIOs are + * being used to bring out signals to the processor then only platform + * data configuration is needed for MAX98090 and processor GPIOs should + * be configured using snd_soc_jack_add_gpios() instead. + * + * If no jack is supplied detection will be disabled. + */ +int max98090_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + dev_info(codec->dev, "max98090_mic_detect\n"); + + max98090->jack = jack; + if (jack) { + snd_soc_update_bits(codec, M98090_REG_IRQ_ENABLE, + M98090_IRQ_IJDET_MASK, + 1 << M98090_IRQ_IJDET_SHIFT); + } else { + snd_soc_update_bits(codec, M98090_REG_IRQ_ENABLE, + M98090_IRQ_IJDET_MASK, + 0); + } + + /* Send an initial empty report */ + snd_soc_jack_report(max98090->jack, 0, + SND_JACK_HEADSET | SND_JACK_BTN_0); + + return 0; +} +EXPORT_SYMBOL_GPL(max98090_mic_detect); + +#define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max98090_dai_ops = { + .set_sysclk = max98090_dai_set_sysclk, + .set_fmt = max98090_dai_set_fmt, + .set_tdm_slot = max98090_set_tdm_slot, + .hw_params = max98090_dai_hw_params, + .digital_mute = max98090_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98090_dai[] = { +{ + .name = "HiFi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98090_RATES, + .formats = MAX98090_FORMATS, + }, + .ops = &max98090_dai_ops, +} +}; + +static void max98090_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_pdata *pdata = max98090->pdata; + + if (!pdata) { + dev_err(codec->dev, "No platform data\n"); + return; + } + +} + +static int max98090_probe(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + struct max98090_cdata *cdata; + int ret = 0; + + dev_info(codec->dev, "max98090_probe\n"); + + max98090->codec = codec; + + codec->control_data = max98090->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Reset the codec, the DSP core, and disable all interrupts */ + max98090_reset(max98090); + + /* Initialize private data */ + + max98090->sysclk = (unsigned)-1; + + cdata = &max98090->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + max98090->lin_state = 0; + max98090->mic1pre = 0; + max98090->mic2pre = 0; + max98090->extmic_mux = 0; + + ret = snd_soc_read(codec, M98090_REG_REV_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read device revision: %d\n", + ret); + goto err_access; + } + + if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { + max98090->devtype = MAX98090; + dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); + } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { + max98090->devtype = MAX98091; + dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); + } else { + max98090->devtype = MAX98090; + dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); + } + + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; + + INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + + /* Enable jack detection */ + snd_soc_write(codec, M98090_REG_CFG_JACK, + M98090_JDETEN_MASK | M98090_JDEB_25MS); + + /* Register for interrupts */ + dev_dbg(codec->dev, "irq = %d\n", max98090->irq); + + ret = request_threaded_irq(max98090->irq, NULL, + max98090_interrupt, IRQF_TRIGGER_FALLING, + "max98090_interrupt", codec); + if (ret < 0) { + dev_err(codec->dev, "request_irq failed: %d\n", + ret); + } + + /* + * Clear any old interrupts. + * An old interrupt ocurring prior to installing the ISR + * can keep a new interrupt from generating a trigger. + */ + snd_soc_read(codec, M98090_REG_IRQ_STATUS); + + /* High Performance is default */ + snd_soc_update_bits(codec, M98090_REG_DAC_CFG, + M98090_DACHP_MASK, + 1 << M98090_DACHP_SHIFT); + snd_soc_update_bits(codec, M98090_REG_DAC_CFG, + M98090_DAC_PERFMODE_MASK, + 0 << M98090_DAC_PERFMODE_SHIFT); + snd_soc_update_bits(codec, M98090_REG_ADC_CFG, + M98090_ADCHP_MASK, + 1 << M98090_ADCHP_SHIFT); + + /* Turn on VCM bandgap reference */ + snd_soc_write(codec, M98090_REG_BIAS_CNTL, + M98090_VCM_MODE_MASK); + + max98090_handle_pdata(codec); + + max98090_add_widgets(codec); + +err_access: + return ret; +} + +static int max98090_remove(struct snd_soc_codec *codec) +{ + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&max98090->jack_work); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98090 = { + .probe = max98090_probe, + .remove = max98090_remove, + .set_bias_level = max98090_set_bias_level, +}; + +static const struct regmap_config max98090_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX98090_MAX_REGISTER, + .reg_defaults = max98090_reg, + .num_reg_defaults = ARRAY_SIZE(max98090_reg), + .volatile_reg = max98090_volatile_register, + .readable_reg = max98090_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98090_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98090_priv *max98090; + int ret; + + pr_info("max98090_i2c_probe\n"); + + max98090 = kzalloc(sizeof(struct max98090_priv), GFP_KERNEL); + if (max98090 == NULL) + return -ENOMEM; + + max98090->devtype = id->driver_data; + i2c_set_clientdata(i2c, max98090); + max98090->control_data = i2c; + max98090->pdata = i2c->dev.platform_data; + max98090->irq = i2c->irq; + + max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + if (IS_ERR(max98090->regmap)) { + ret = PTR_ERR(max98090->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max98090, max98090_dai, + ARRAY_SIZE(max98090_dai)); + if (ret < 0) + regmap_exit(max98090->regmap); + +err_enable: + return ret; +} + +static int __devexit max98090_i2c_remove(struct i2c_client *client) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); + snd_soc_unregister_codec(&client->dev); + regmap_exit(max98090->regmap); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static int max98090_runtime_resume(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, false); + + max98090_reset(max98090); + + regcache_sync(max98090->regmap); + + return 0; +} + +static int max98090_runtime_suspend(struct device *dev) +{ + struct max98090_priv *max98090 = dev_get_drvdata(dev); + + regcache_cache_only(max98090->regmap, true); + + return 0; +} + +static struct dev_pm_ops max98090_pm = { + SET_RUNTIME_PM_OPS(max98090_runtime_suspend, + max98090_runtime_resume, NULL) +}; + +static const struct i2c_device_id max98090_i2c_id[] = { + { "max98090", MAX98090 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); + +static struct i2c_driver max98090_i2c_driver = { + .driver = { + .name = "max98090", + .owner = THIS_MODULE, + .pm = &max98090_pm, + }, + .probe = max98090_i2c_probe, + .remove = __devexit_p(max98090_i2c_remove), + .id_table = max98090_i2c_id, +}; + +module_i2c_driver(max98090_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98090 driver"); +MODULE_AUTHOR("Peter Hsiang, Jesse Marroqin, Jerry Wong"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h new file mode 100755 index 0000000..4d48381 --- /dev/null +++ b/sound/soc/codecs/max98090.h @@ -0,0 +1,1537 @@ +/* + * max98090.h -- MAX98090 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98090_H +#define _MAX98090_H + +#include <linux/version.h> + +/* One can override the Linux version here with an explicit version number */ +#define M98090_LINUX_VERSION LINUX_VERSION_CODE + +/* + * MAX98090 Register Definitions + */ + +#define M98090_REG_SW_RESET 0x00 +#define M98090_REG_IRQ_STATUS 0x01 +#define M98090_REG_JACK_STATUS 0x02 +#define M98090_REG_IRQ_ENABLE 0x03 +#define M98090_REG_QCFG_SYS_CLK 0x04 +#define M98090_REG_QCFG_RATE 0x05 +#define M98090_REG_QCFG_DAI 0x06 +#define M98090_REG_QCFG_DAC 0x07 +#define M98090_REG_QCFG_MIC_PATH 0x08 +#define M98090_REG_QCFG_LINE_PATH 0x09 +#define M98090_REG_QCFG_MIC_LOOP 0x0A +#define M98090_REG_QCFG_LINE_LOOP 0x0B +#define M98090_REG_RESERVED 0x0C +#define M98090_REG_CFG_INPUT 0x0D +#define M98090_REG_LVL_LINE 0x0E +#define M98090_REG_CFG_LINE 0x0F +#define M98090_REG_LVL_MIC1 0x10 +#define M98090_REG_LVL_MIC2 0x11 +#define M98090_REG_MIC_BIAS 0x12 +#define M98090_REG_MIC_CFG1 0x13 +#define M98090_REG_MIC_CFG2 0x14 +#define M98090_REG_MIX_ADC_L 0x15 +#define M98090_REG_MIX_ADC_R 0x16 +#define M98090_REG_LVL_ADC_L 0x17 +#define M98090_REG_LVL_ADC_R 0x18 +#define M98090_REG_LVL_BIQUAD 0x19 +#define M98090_REG_LVL_SIDETONE 0x1A +#define M98090_REG_SYS_CLOCK 0x1B +#define M98090_REG_CLOCK_MODE 0x1C +#define M98090_REG_CLOCK_DAI1_NI_HI 0x1D +#define M98090_REG_CLOCK_DAI2_NI_LO 0x1E +#define M98090_REG_CLOCK_DAI3_MI_HI 0x1F +#define M98090_REG_CLOCK_DAI4_MI_LO 0x20 +#define M98090_REG_CLOCK_MAS_MODE 0x21 +#define M98090_REG_DAI_INTERFACE_FORMAT 0x22 +#define M98090_REG_DAI_TDM_CONTROL 0x23 +#define M98090_REG_DAI_TDM_FORMAT 0x24 +#define M98090_REG_DAI_IOCFG 0x25 +#define M98090_REG_DAI_FILTERS 0x26 +#define M98090_REG_DAI_LVL 0x27 +#define M98090_REG_DAI_LVL_EQ 0x28 +#define M98090_REG_MIX_HP_LEFT 0x29 +#define M98090_REG_MIX_HP_RIGHT 0x2A +#define M98090_REG_MIX_HP_CNTL 0x2B +#define M98090_REG_LVL_HP_LEFT 0x2C +#define M98090_REG_LVL_HP_RIGHT 0x2D +#define M98090_REG_MIX_SPK_LEFT 0x2E +#define M98090_REG_MIX_SPK_RIGHT 0x2F +#define M98090_REG_MIX_SPK_CNTL 0x30 +#define M98090_REG_LVL_SPK_LEFT 0x31 +#define M98090_REG_LVL_SPK_RIGHT 0x32 +#define M98090_REG_ALC_TIMING 0x33 +#define M98090_REG_ALC_CMPR 0x34 +#define M98090_REG_ALC_EXP 0x35 +#define M98090_REG_LVL_ALC 0x36 +#define M98090_REG_MIX_RCV_LEFT 0x37 +#define M98090_REG_MIX_RCV_CTRL_LEFT 0x38 +#define M98090_REG_LVL_RCV_LEFT 0x39 +#define M98090_REG_MIX_RCV_RIGHT 0x3A +#define M98090_REG_MIX_RCV_CNTL_RIGHT 0x3B +#define M98090_REG_LVL_RCV_RIGHT 0x3C +#define M98090_REG_CFG_JACK 0x3D +#define M98090_REG_PWR_EN_IN 0x3E +#define M98090_REG_PWR_EN_OUT 0x3F +#define M98090_REG_CFG_LVL 0x40 +#define M98090_REG_DSP_EQ_EN 0x41 +#define M98090_REG_BIAS_CNTL 0x42 +#define M98090_REG_DAC_CFG 0x43 +#define M98090_REG_ADC_CFG 0x44 +#define M98090_REG_PWR_SYS 0x45 +#define M98090_REG_EQ_BASE 0x46 +#define M98090_REG_BIQUAD_BASE 0xAF +#define M98090_REG_DMIC3_VOLUME 0xBE +#define M98090_REG_DMIC4_VOLUME 0xBF +#define M98090_REG_DMIC34_BQ_PREATTEN 0xC0 +#define M98090_REG_RECORD_TDM_SLOT 0xC1 +#define M98090_REG_SAMPLE_RATE 0xC2 +#define M98090_REG_DMIC34_BIQUAD_BASE 0xC3 +#define M98090_REG_REV_ID 0xFF + +#define M98090_REG_CNT (0xFF+1) +#define MAX98090_MAX_REGISTER 0xFF + +/* MAX98090 Register Bit Fields */ + +/* + * M98090_REG_SW_RESET + */ +#define M98090_SWRESET_MASK (1<<7) +#define M98090_SWRESET_SHIFT 7 +#define M98090_SWRESET_WIDTH 1 + +/* + * M98090_REG_IRQ_STATUS + */ +#define M98090_IRQ_CLD_MASK (1<<7) +#define M98090_IRQ_CLD_SHIFT 7 +#define M98090_IRQ_CLD_WIDTH 1 +#define M98090_IRQ_SLD_MASK (1<<6) +#define M98090_IRQ_SLD_SHIFT 6 +#define M98090_IRQ_SLD_WIDTH 1 +#define M98090_IRQ_ULK_MASK (1<<5) +#define M98090_IRQ_ULK_SHIFT 5 +#define M98090_IRQ_ULK_WIDTH 1 +#define M98090_IRQ_JDET_MASK (1<<2) +#define M98090_IRQ_JDET_SHIFT 2 +#define M98090_IRQ_JDET_WIDTH 1 +#define M98090_IRQ_ALCACT_MASK (1<<1) +#define M98090_IRQ_ALCACT_SHIFT 1 +#define M98090_IRQ_ALCACT_WIDTH 1 +#define M98090_IRQ_ALCCLP_MASK (1<<0) +#define M98090_IRQ_ALCCLP_SHIFT 0 +#define M98090_IRQ_ALCCLP_WIDTH 1 + +/* + * M98090_REG_JACK_STATUS + */ +#define M98090_LSNS_MASK (1<<2) +#define M98090_LSNS_SHIFT 2 +#define M98090_LSNS_WIDTH 1 +#define M98090_JKSNS_MASK (1<<1) +#define M98090_JKSNS_SHIFT 1 +#define M98090_JKSNS_WIDTH 1 + +/* + * M98090_REG_IRQ_ENABLE + */ +#define M98090_IRQ_ICLD_MASK (1<<7) +#define M98090_IRQ_ICLD_SHIFT 7 +#define M98090_IRQ_ICLD_WIDTH 1 +#define M98090_IRQ_ISLD_MASK (1<<6) +#define M98090_IRQ_ISLD_SHIFT 6 +#define M98090_IRQ_ISLD_WIDTH 1 +#define M98090_IRQ_IULK_MASK (1<<5) +#define M98090_IRQ_IULK_SHIFT 5 +#define M98090_IRQ_IULK_WIDTH 1 +#define M98090_IRQ_IJDET_MASK (1<<2) +#define M98090_IRQ_IJDET_SHIFT 2 +#define M98090_IRQ_IJDET_WIDTH 1 +#define M98090_IRQ_IALCACT_MASK (1<<1) +#define M98090_IRQ_IALCACT_SHIFT 1 +#define M98090_IRQ_IALCACT_WIDTH 1 +#define M98090_IRQ_IALCCLP_MASK (1<<0) +#define M98090_IRQ_IALCCLP_SHIFT 0 +#define M98090_IRQ_IALCCLP_WIDTH 1 + +/* + * M98090_REG_QCFG_SYS_CLK + */ +#define M98090_CLK_26M_MASK (1<<7) +#define M98090_CLK_26M_SHIFT 7 +#define M98090_CLK_26M_WIDTH 1 +#define M98090_CLK_19P2M_MASK (1<<6) +#define M98090_CLK_19P2M_SHIFT 6 +#define M98090_CLK_19P2M_WIDTH 1 +#define M98090_CLK_13M_MASK (1<<5) +#define M98090_CLK_13M_SHIFT 5 +#define M98090_CLK_13M_WIDTH 1 +#define M98090_CLK_12P288M_MASK (1<<4) +#define M98090_CLK_12P288M_SHIFT 4 +#define M98090_CLK_12P288M_WIDTH 1 +#define M98090_CLK_12M_MASK (1<<3) +#define M98090_CLK_12M_SHIFT 3 +#define M98090_CLK_12M_WIDTH 1 +#define M98090_CLK_11P2896M_MASK (1<<2) +#define M98090_CLK_11P2896M_SHIFT 2 +#define M98090_CLK_11P2896M_WIDTH 1 +#define M98090_CLK_256FS_MASK (1<<0) +#define M98090_CLK_256FS_SHIFT 0 +#define M98090_CLK_256FS_WIDTH 1 +#define M98090_CLK_ALL_SHIFT 0 +#define M98090_CLK_ALL_WIDTH 8 +#define M98090_CLK_ALL_NUM (1<<M98090_CLK_ALL_WIDTH) + +/* + * M98090_REG_QCFG_RATE + */ +#define M98090_SR_96K_MASK (1<<5) +#define M98090_SR_96K_SHIFT 5 +#define M98090_SR_96K_WIDTH 1 +#define M98090_SR_32K_MASK (1<<4) +#define M98090_SR_32K_SHIFT 4 +#define M98090_SR_32K_WIDTH 1 +#define M98090_SR_48K_MASK (1<<3) +#define M98090_SR_48K_SHIFT 3 +#define M98090_SR_48K_WIDTH 1 +#define M98090_SR_44K1_MASK (1<<2) +#define M98090_SR_44K1_SHIFT 2 +#define M98090_SR_44K1_WIDTH 1 +#define M98090_SR_16K_MASK (1<<1) +#define M98090_SR_16K_SHIFT 1 +#define M98090_SR_16K_WIDTH 1 +#define M98090_SR_8K_MASK (1<<0) +#define M98090_SR_8K_SHIFT 0 +#define M98090_SR_8K_WIDTH 1 +#define M98090_SR_MASK 0x3F +#define M98090_SR_ALL_SHIFT 0 +#define M98090_SR_ALL_WIDTH 8 +#define M98090_SR_ALL_NUM (1<<M98090_SR_ALL_WIDTH) + +/* + * M98090_REG_QCFG_DAI + */ +#define M98090_DAI_RJ_M_MASK (1<<5) +#define M98090_DAI_RJ_M_SHIFT 5 +#define M98090_DAI_RJ_M_WIDTH 1 +#define M98090_DAI_RJ_S_MASK (1<<4) +#define M98090_DAI_RJ_S_SHIFT 4 +#define M98090_DAI_RJ_S_WIDTH 1 +#define M98090_DAI_LJ_M_MASK (1<<3) +#define M98090_DAI_LJ_M_SHIFT 3 +#define M98090_DAI_LJ_M_WIDTH 1 +#define M98090_DAI_LJ_S_MASK (1<<2) +#define M98090_DAI_LJ_S_SHIFT 2 +#define M98090_DAI_LJ_S_WIDTH 1 +#define M98090_DAI_I2S_M_MASK (1<<1) +#define M98090_DAI_I2S_M_SHIFT 1 +#define M98090_DAI_I2S_M_WIDTH 1 +#define M98090_DAI_I2S_S_MASK (1<<0) +#define M98090_DAI_I2S_S_SHIFT 0 +#define M98090_DAI_I2S_S_WIDTH 1 +#define M98090_DAI_ALL_SHIFT 0 +#define M98090_DAI_ALL_WIDTH 8 +#define M98090_DAI_ALL_NUM (1<<M98090_DAI_ALL_WIDTH) + +/* + * M98090_REG_QCFG_DAC + */ +#define M98090_DIG2_HP_MASK (1<<7) +#define M98090_DIG2_HP_SHIFT 7 +#define M98090_DIG2_HP_WIDTH 1 +#define M98090_DIG2_EAR_MASK (1<<6) +#define M98090_DIG2_EAR_SHIFT 6 +#define M98090_DIG2_EAR_WIDTH 1 +#define M98090_DIG2_SPK_MASK (1<<5) +#define M98090_DIG2_SPK_SHIFT 5 +#define M98090_DIG2_SPK_WIDTH 1 +#define M98090_DIG2_LOUT_MASK (1<<4) +#define M98090_DIG2_LOUT_SHIFT 4 +#define M98090_DIG2_LOUT_WIDTH 1 +#define M98090_DIG2_ALL_SHIFT 0 +#define M98090_DIG2_ALL_WIDTH 8 +#define M98090_DIG2_ALL_NUM (1<<M98090_DIG2_ALL_WIDTH) + +/* + * M98090_REG_QCFG_MIC_PATH + */ +#define M98090_IN12_MIC1_MASK (1<<7) +#define M98090_IN12_MIC1_SHIFT 7 +#define M98090_IN12_MIC1_WIDTH 1 +#define M98090_IN34_MIC2_MASK (1<<6) +#define M98090_IN34_MIC2_SHIFT 6 +#define M98090_IN34_MIC2_WIDTH 1 +#define M98090_IN56_MIC1_MASK (1<<5) +#define M98090_IN56_MIC1_SHIFT 5 +#define M98090_IN56_MIC1_WIDTH 1 +#define M98090_IN56_MIC2_MASK (1<<4) +#define M98090_IN56_MIC2_SHIFT 4 +#define M98090_IN56_MIC2_WIDTH 1 +#define M98090_IN12_DADC_MASK (1<<3) +#define M98090_IN12_DADC_SHIFT 3 +#define M98090_IN12_DADC_WIDTH 1 +#define M98090_IN34_DADC_MASK (1<<2) +#define M98090_IN34_DADC_SHIFT 2 +#define M98090_IN34_DADC_WIDTH 1 +#define M98090_IN56_DADC_MASK (1<<1) +#define M98090_IN56_DADC_SHIFT 1 +#define M98090_IN56_DADC_WIDTH 1 +#define M98090_MIC_ALL_SHIFT 0 +#define M98090_MIC_ALL_WIDTH 8 +#define M98090_MIC_ALL_NUM (1<<M98090_MIC_ALL_WIDTH) + +/* + * M98090_REG_QCFG_LINE_PATH + */ +#define M98090_IN12S_AB_MASK (1<<7) +#define M98090_IN12S_AB_SHIFT 7 +#define M98090_IN12S_AB_WIDTH 1 +#define M98090_IN34S_AB_MASK (1<<6) +#define M98090_IN34S_AB_SHIFT 6 +#define M98090_IN34S_AB_WIDTH 1 +#define M98090_IN56S_AB_MASK (1<<5) +#define M98090_IN56S_AB_SHIFT 5 +#define M98090_IN56S_AB_WIDTH 1 +#define M98090_IN34D_A_MASK (1<<4) +#define M98090_IN34D_A_SHIFT 4 +#define M98090_IN34D_A_WIDTH 1 +#define M98090_IN56D_B_MASK (1<<3) +#define M98090_IN56D_B_SHIFT 3 +#define M98090_IN56D_B_WIDTH 1 +#define M98090_LINE_ALL_SHIFT 0 +#define M98090_LINE_ALL_WIDTH 8 +#define M98090_LINE_ALL_NUM (1<<M98090_LINE_ALL_WIDTH) + +/* + * M98090_REG_QCFG_MIC_LOOP + */ +#define M98090_IN12_M1HPL_MASK (1<<7) +#define M98090_IN12_M1HPL_SHIFT 7 +#define M98090_IN12_M1HPL_WIDTH 1 +#define M98090_IN12_M1SPKL_MASK (1<<6) +#define M98090_IN12_M1SPKL_SHIFT 6 +#define M98090_IN12_M1SPKL_WIDTH 1 +#define M98090_IN12_M1EAR_MASK (1<<5) +#define M98090_IN12_M1EAR_SHIFT 5 +#define M98090_IN12_M1EAR_WIDTH 1 +#define M98090_IN12_M1LOUTL_MASK (1<<4) +#define M98090_IN12_M1LOUTL_SHIFT 4 +#define M98090_IN12_M1LOUTL_WIDTH 1 +#define M98090_IN34_M2HPR_MASK (1<<3) +#define M98090_IN34_M2HPR_SHIFT 3 +#define M98090_IN34_M2HPR_WIDTH 1 +#define M98090_IN34_M2SPKR_MASK (1<<2) +#define M98090_IN34_M2SPKR_SHIFT 2 +#define M98090_IN34_M2SPKR_WIDTH 1 +#define M98090_IN34_M2EAR_MASK (1<<1) +#define M98090_IN34_M2EAR_SHIFT 1 +#define M98090_IN34_M2EAR_WIDTH 1 +#define M98090_IN34_M2LOUTR_MASK (1<<0) +#define M98090_IN34_M2LOUTR_SHIFT 0 +#define M98090_IN34_M2LOUTR_WIDTH 1 +#define M98090_AMIC_ALL_SHIFT 0 +#define M98090_AMIC_ALL_WIDTH 8 +#define M98090_AMIC_ALL_NUM (1<<M98090_AMIC_ALL_WIDTH) + +/* + * M98090_REG_QCFG_LINE_LOOP + */ +#define M98090_IN12S_ABHP_MASK (1<<7) +#define M98090_IN12S_ABHP_SHIFT 7 +#define M98090_IN12S_ABHP_WIDTH 1 +#define M98090_IN34D_ASPKL_MASK (1<<6) +#define M98090_IN34D_ASPKL_SHIFT 6 +#define M98090_IN34D_ASPKL_WIDTH 1 +#define M98090_IN34D_AEAR_MASK (1<<5) +#define M98090_IN34D_AEAR_SHIFT 5 +#define M98090_IN34D_AEAR_WIDTH 1 +#define M98090_IN12S_ABLOUT_MASK (1<<4) +#define M98090_IN12S_ABLOUT_SHIFT 4 +#define M98090_IN12S_ABLOUT_WIDTH 1 +#define M98090_IN34S_ABHP_MASK (1<<3) +#define M98090_IN34S_ABHP_SHIFT 3 +#define M98090_IN34S_ABHP_WIDTH 1 +#define M98090_IN56D_BSPKR_MASK (1<<2) +#define M98090_IN56D_BSPKR_SHIFT 2 +#define M98090_IN56D_BSPKR_WIDTH 1 +#define M98090_IN56D_BEAR_MASK (1<<1) +#define M98090_IN56D_BEAR_SHIFT 1 +#define M98090_IN56D_BEAR_WIDTH 1 +#define M98090_IN34S_ABLOUT_MASK (1<<0) +#define M98090_IN34S_ABLOUT_SHIFT 0 +#define M98090_IN34S_ABLOUT_WIDTH 1 +#define M98090_ALIN_ALL_SHIFT 0 +#define M98090_ALIN_ALL_WIDTH 8 +#define M98090_ALIN_ALL_NUM (1<<M98090_ALIN_ALL_WIDTH) + +/* + * M98090_REG_RESERVED + */ + +/* + * M98090_REG_CFG_INPUT + */ +#define M98090_IN34DIFF_MASK (1<<7) +#define M98090_IN34DIFF_SHIFT 7 +#define M98090_IN34DIFF_WIDTH 1 +#define M98090_IN56DIFF_MASK (1<<6) +#define M98090_IN56DIFF_SHIFT 6 +#define M98090_IN56DIFF_WIDTH 1 +#define M98090_IN1SEEN_MASK (1<<5) +#define M98090_IN1SEEN_SHIFT 5 +#define M98090_IN1SEEN_WIDTH 1 +#define M98090_IN2SEEN_MASK (1<<4) +#define M98090_IN2SEEN_SHIFT 4 +#define M98090_IN2SEEN_WIDTH 1 +#define M98090_IN3SEEN_MASK (1<<3) +#define M98090_IN3SEEN_SHIFT 3 +#define M98090_IN3SEEN_WIDTH 1 +#define M98090_IN4SEEN_MASK (1<<2) +#define M98090_IN4SEEN_SHIFT 2 +#define M98090_IN4SEEN_WIDTH 1 +#define M98090_IN5SEEN_MASK (1<<1) +#define M98090_IN5SEEN_SHIFT 1 +#define M98090_IN5SEEN_WIDTH 1 +#define M98090_IN6SEEN_MASK (1<<0) +#define M98090_IN6SEEN_SHIFT 0 +#define M98090_IN6SEEN_WIDTH 1 + +/* + * M98090_REG_LVL_LINE + */ +#define M98090_MIXG135_MASK (1<<7) +#define M98090_MIXG135_SHIFT 7 +#define M98090_MIXG135_WIDTH 1 +#define M98090_MIXT246_MASK (1<<6) +#define M98090_MIXT246_SHIFT 6 +#define M98090_MIXT246_WIDTH 1 +#define M98090_LINAPGA_MASK (7<<3) +#define M98090_LINAPGA_SHIFT 3 +#define M98090_LINAPGA_WIDTH 3 +#define M98090_LINAPGA_NUM 6 +#define M98090_LINBPGA_MASK (7<<0) +#define M98090_LINBPGA_SHIFT 0 +#define M98090_LINBPGA_WIDTH 3 +#define M98090_LINBPGA_NUM 6 + +/* + * M98090_REG_CFG_LINE + */ +#define M98090_EXTBUFA_MASK (1<<7) +#define M98090_EXTBUFA_SHIFT 7 +#define M98090_EXTBUFA_WIDTH 1 +#define M98090_EXTBUFA_NUM (1<<M98090_EXTBUFA_WIDTH) +#define M98090_EXTBUFB_MASK (1<<6) +#define M98090_EXTBUFB_SHIFT 6 +#define M98090_EXTBUFB_WIDTH 1 +#define M98090_EXTBUFB_NUM (1<<M98090_EXTBUFB_WIDTH) +#define M98090_EXTMIC_MASK (3<<0) +#define M98090_EXTMIC_SHIFT 0 +#define M98090_EXTMIC1_SHIFT 0 +#define M98090_EXTMIC2_SHIFT 1 +#define M98090_EXTMIC_WIDTH 2 +#define M98090_EXTMIC_NONE (0<<0) +#define M98090_EXTMIC_MIC1 (1<<0) +#define M98090_EXTMIC_MIC2 (2<<0) + +/* + * M98090_REG_LVL_MIC1 + */ +#define M98090_MIC_PA1EN_MASK (3<<5) +#define M98090_MIC_PA1EN_SHIFT 5 +#define M98090_MIC_PA1EN_WIDTH 2 +#define M98090_MIC_PA1EN_NUM 3 +#define M98090_MIC_PGAM1_MASK (31<<0) +#define M98090_MIC_PGAM1_SHIFT 0 +#define M98090_MIC_PGAM1_WIDTH 5 +#define M98090_MIC_PGAM1_NUM 21 + +/* + * M98090_REG_LVL_MIC2 + */ +#define M98090_MIC_PA2EN_MASK (3<<5) +#define M98090_MIC_PA2EN_SHIFT 5 +#define M98090_MIC_PA2EN_WIDTH 2 +#define M98090_MIC_PA2EN_NUM 3 +#define M98090_MIC_PGAM2_MASK (31<<0) +#define M98090_MIC_PGAM2_SHIFT 0 +#define M98090_MIC_PGAM2_WIDTH 5 +#define M98090_MIC_PGAM2_NUM 21 + +/* + * M98090_REG_MIC_BIAS + */ +#define M98090_MBVSEL_MASK (3<<0) +#define M98090_MBVSEL_SHIFT 0 +#define M98090_MBVSEL_WIDTH 2 +#define M98090_MBVSEL_2V8 (3<<0) +#define M98090_MBVSEL_2V55 (2<<0) +#define M98090_MBVSEL_2V4 (1<<0) +#define M98090_MBVSEL_2V2 (0<<0) + +/* + * M98090_REG_MIC_CFG1 + */ +#define M98090_MICCLK_MASK (7<<4) +#define M98090_MICCLK_SHIFT 4 +#define M98090_MICCLK_WIDTH 3 +#define M98090_DIGMIC4_MASK (1<<3) +#define M98090_DIGMIC4_SHIFT 3 +#define M98090_DIGMIC4_WIDTH 1 +#define M98090_DIGMIC4_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DIGMIC3_MASK (1<<2) +#define M98090_DIGMIC3_SHIFT 2 +#define M98090_DIGMIC3_WIDTH 1 +#define M98090_DIGMIC3_NUM (1<<M98090_DIGMIC3_WIDTH) +#define M98090_DIGMICR_MASK (1<<1) +#define M98090_DIGMICR_SHIFT 1 +#define M98090_DIGMICR_WIDTH 1 +#define M98090_DIGMICR_NUM (1<<M98090_DIGMICR_WIDTH) +#define M98090_DIGMICL_MASK (1<<0) +#define M98090_DIGMICL_SHIFT 0 +#define M98090_DIGMICL_WIDTH 1 +#define M98090_DIGMICL_NUM (1<<M98090_DIGMICL_WIDTH) + +/* + * M98090_REG_MIC_CFG2 + */ +#define M98090_DMIC_COMP_MASK (15<<4) +#define M98090_DMIC_COMP_SHIFT 4 +#define M98090_DMIC_COMP_WIDTH 4 +#define M98090_DMIC_COMP_NUM (1<<M98090_DMIC_COMP_WIDTH) +#define M98090_DMIC_FREQ_MASK (3<<0) +#define M98090_DMIC_FREQ_SHIFT 0 +#define M98090_DMIC_FREQ_WIDTH 2 + +/* + * M98090_REG_MIX_ADC_L + */ +#define M98090_MIXADL_MIC2_MASK (1<<6) +#define M98090_MIXADL_MIC2_SHIFT 6 +#define M98090_MIXADL_MIC2_WIDTH 1 +#define M98090_MIXADL_MIC1_MASK (1<<5) +#define M98090_MIXADL_MIC1_SHIFT 5 +#define M98090_MIXADL_MIC1_WIDTH 1 +#define M98090_MIXADL_LINEB_MASK (1<<4) +#define M98090_MIXADL_LINEB_SHIFT 4 +#define M98090_MIXADL_LINEB_WIDTH 1 +#define M98090_MIXADL_LINEA_MASK (1<<3) +#define M98090_MIXADL_LINEA_SHIFT 3 +#define M98090_MIXADL_LINEA_WIDTH 1 +#define M98090_MIXADL_IN65DIFF_MASK (1<<2) +#define M98090_MIXADL_IN65DIFF_SHIFT 2 +#define M98090_MIXADL_IN65DIFF_WIDTH 1 +#define M98090_MIXADL_IN34DIFF_MASK (1<<1) +#define M98090_MIXADL_IN34DIFF_SHIFT 1 +#define M98090_MIXADL_IN34DIFF_WIDTH 1 +#define M98090_MIXADL_IN12DIFF_MASK (1<<0) +#define M98090_MIXADL_IN12DIFF_SHIFT 0 +#define M98090_MIXADL_IN12DIFF_WIDTH 1 +#define M98090_MIXADL_MASK (255<<0) +#define M98090_MIXADL_SHIFT 0 +#define M98090_MIXADL_WIDTH 8 + +/* + * M98090_REG_MIX_ADC_R + */ +#define M98090_MIXADR_MIC2_MASK (1<<6) +#define M98090_MIXADR_MIC2_SHIFT 6 +#define M98090_MIXADR_MIC2_WIDTH 1 +#define M98090_MIXADR_MIC1_MASK (1<<5) +#define M98090_MIXADR_MIC1_SHIFT 5 +#define M98090_MIXADR_MIC1_WIDTH 1 +#define M98090_MIXADR_LINEB_MASK (1<<4) +#define M98090_MIXADR_LINEB_SHIFT 4 +#define M98090_MIXADR_LINEB_WIDTH 1 +#define M98090_MIXADR_LINEA_MASK (1<<3) +#define M98090_MIXADR_LINEA_SHIFT 3 +#define M98090_MIXADR_LINEA_WIDTH 1 +#define M98090_MIXADR_IN65DIFF_MASK (1<<2) +#define M98090_MIXADR_IN65DIFF_SHIFT 2 +#define M98090_MIXADR_IN65DIFF_WIDTH 1 +#define M98090_MIXADR_IN34DIFF_MASK (1<<1) +#define M98090_MIXADR_IN34DIFF_SHIFT 1 +#define M98090_MIXADR_IN34DIFF_WIDTH 1 +#define M98090_MIXADR_IN12DIFF_MASK (1<<0) +#define M98090_MIXADR_IN12DIFF_SHIFT 0 +#define M98090_MIXADR_IN12DIFF_WIDTH 1 +#define M98090_MIXADR_MASK (255<<0) +#define M98090_MIXADR_SHIFT 0 +#define M98090_MIXADR_WIDTH 8 + +/* + * M98090_REG_LVL_ADC_L + */ +#define M98090_ADC_AVLG_MASK (7<<4) +#define M98090_ADC_AVLG_SHIFT 4 +#define M98090_ADC_AVLG_WIDTH 3 +#define M98090_ADC_AVLG_NUM (1<<M98090_ADC_AVLG_WIDTH) +#define M98090_ADC_AVL_MASK (15<<0) +#define M98090_ADC_AVL_SHIFT 0 +#define M98090_ADC_AVL_WIDTH 4 +#define M98090_ADC_AVL_NUM (1<<M98090_ADC_AVL_WIDTH) + +/* + * M98090_REG_LVL_ADC_R + */ +#define M98090_ADC_AVRG_MASK (7<<4) +#define M98090_ADC_AVRG_SHIFT 4 +#define M98090_ADC_AVRG_WIDTH 3 +#define M98090_ADC_AVRG_NUM (1<<M98090_ADC_AVRG_WIDTH) +#define M98090_ADC_AVR_MASK (15<<0) +#define M98090_ADC_AVR_SHIFT 0 +#define M98090_ADC_AVR_WIDTH 4 +#define M98090_ADC_AVR_NUM (1<<M98090_ADC_AVR_WIDTH) + +/* + * M98090_REG_LVL_BIQUAD + */ +#define M98090_AVBQ_MASK (15<<0) +#define M98090_AVBQ_SHIFT 0 +#define M98090_AVBQ_WIDTH 4 +#define M98090_AVBQ_NUM (1<<M98090_AVBQ_WIDTH) + +/* + * M98090_REG_LVL_SIDETONE + */ +#define M98090_DSTS_MASK (3<<6) +#define M98090_DSTS_SHIFT 6 +#define M98090_DSTS_WIDTH 2 +#define M98090_DVST_MASK (31<<0) +#define M98090_DVST_SHIFT 0 +#define M98090_DVST_WIDTH 5 +#define M98090_DVST_NUM 31 + +/* + * M98090_REG_SYS_CLOCK + */ +#define M98090_CLK_PSCLK_MASK (3<<4) +#define M98090_CLK_PSCLK_SHIFT 4 +#define M98090_CLK_PSCLK_WIDTH 2 +#define M98090_CLK_PSCLK_DISABLED (0<<4) +#define M98090_CLK_PSCLK_DIV1 (1<<4) +#define M98090_CLK_PSCLK_DIV2 (2<<4) +#define M98090_CLK_PSCLK_DIV4 (3<<4) + +/* + * M98090_REG_CLOCK_MODE + */ +#define M98090_CLK_FREQ1_MASK (15<<4) +#define M98090_CLK_FREQ1_SHIFT 4 +#define M98090_CLK_FREQ1_WIDTH 4 +#define M98090_CLK_USE_M1_MASK (1<<0) +#define M98090_CLK_USE_M1_SHIFT 0 +#define M98090_CLK_USE_M1_WIDTH 1 +#define M98090_CLK_USE_M1_NUM (1<<M98090_CLK_USE_M1_WIDTH) + +/* + * M98090_REG_CLOCK_DAI1_NI_HI + */ +#define M98090_CLK_NI_HI_MASK (127<<0) +#define M98090_CLK_NI_HI_SHIFT 0 +#define M98090_CLK_NI_HI_WIDTH 7 +#define M98090_CLK_NI_HI_NUM (1<<M98090_CLK_NI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_DAI2_NI_LO + */ +#define M98090_CLK_NI_LO_MASK (255<<0) +#define M98090_CLK_NI_LO_SHIFT 0 +#define M98090_CLK_NI_LO_WIDTH 8 +#define M98090_CLK_NI_LO_NUM (1<<M98090_CLK_NI_LO_WIDTH) + +/* + * M98090_REG_CLOCK_DAI3_MI_HI + */ +#define M98090_CLK_MI_HI_MASK (255<<0) +#define M98090_CLK_MI_HI_SHIFT 0 +#define M98090_CLK_MI_HI_WIDTH 8 +#define M98090_CLK_MI_HI_NUM (1<<M98090_CLK_MI_HI_WIDTH) + +/* + * M98090_REG_CLOCK_DAI4_MI_LO + */ +#define M98090_CLK_MI_LO_MASK (255<<0) +#define M98090_CLK_MI_LO_SHIFT 0 +#define M98090_CLK_MI_LO_WIDTH 8 +#define M98090_CLK_MI_LO_NUM (1<<M98090_CLK_MI_LO_WIDTH) + +/* + * M98090_REG_CLOCK_MAS_MODE + */ +#define M98090_CLK_MAS_MASK (1<<7) +#define M98090_CLK_MAS_SHIFT 7 +#define M98090_CLK_MAS_WIDTH 1 +#define M98090_CLK_BSEL_MASK (1<<0) +#define M98090_CLK_BSEL_SHIFT 0 +#define M98090_CLK_BSEL_WIDTH 1 +#define M98090_CLK_BSEL_32 (1<<0) +#define M98090_CLK_BSEL_48 (2<<0) +#define M98090_CLK_BSEL_64 (3<<0) + +/* + * M98090_REG_DAI_INTERFACE_FORMAT + */ +#define M98090_DAI_RJ_MASK (1<<5) +#define M98090_DAI_RJ_SHIFT 5 +#define M98090_DAI_RJ_WIDTH 1 +#define M98090_DAI_WCI_MASK (1<<4) +#define M98090_DAI_WCI_SHIFT 4 +#define M98090_DAI_WCI_WIDTH 1 +#define M98090_DAI_BCI_MASK (1<<3) +#define M98090_DAI_BCI_SHIFT 3 +#define M98090_DAI_BCI_WIDTH 1 +#define M98090_DAI_DLY_MASK (1<<2) +#define M98090_DAI_DLY_SHIFT 2 +#define M98090_DAI_DLY_WIDTH 1 +#define M98090_DAI_WS_MASK (3<<0) +#define M98090_DAI_WS_SHIFT 0 +#define M98090_DAI_WS_WIDTH 2 +#define M98090_DAI_WS_NUM (1<<M98090_DAI_WS_WIDTH) + +/* + * M98090_REG_DAI_TDM_CONTROL + */ +#define M98090_DAI_FSW_MASK (1<<1) +#define M98090_DAI_FSW_SHIFT 1 +#define M98090_DAI_FSW_WIDTH 1 +#define M98090_DAI_TDM_MASK (1<<0) +#define M98090_DAI_TDM_SHIFT 0 +#define M98090_DAI_TDM_WIDTH 1 +#define M98090_DAI_TDM_NUM (1<<M98090_DAI_TDM_WIDTH) + +/* + * M98090_REG_DAI_TDM_FORMAT + */ +#define M98090_DAI_TDM_SLOTL_MASK (3<<6) +#define M98090_DAI_TDM_SLOTL_SHIFT 6 +#define M98090_DAI_TDM_SLOTL_WIDTH 2 +#define M98090_DAI_TDM_SLOTL_NUM (1<<M98090_DAI_TDM_SLOTL_WIDTH) +#define M98090_DAI_TDM_SLOTR_MASK (3<<4) +#define M98090_DAI_TDM_SLOTR_SHIFT 4 +#define M98090_DAI_TDM_SLOTR_WIDTH 2 +#define M98090_DAI_TDM_SLOTR_NUM (1<<M98090_DAI_TDM_SLOTR_WIDTH) +#define M98090_DAI_TDM_SLOTDLY_MASK (15<<0) +#define M98090_DAI_TDM_SLOTDLY_SHIFT 0 +#define M98090_DAI_TDM_SLOTDLY_WIDTH 4 +#define M98090_DAI_TDM_SLOTDLY_NUM (1<<M98090_DAI_TDM_SLOTDLY_WIDTH) + +/* + * M98090_REG_DAI_IOCFG + */ +#define M98090_DAI_LTEN_MASK (1<<5) +#define M98090_DAI_LTEN_SHIFT 5 +#define M98090_DAI_LTEN_WIDTH 1 +#define M98090_DAI_LTEN_NUM (1<<M98090_DAI_LTEN_WIDTH) +#define M98090_DAI_LBEN_MASK (1<<4) +#define M98090_DAI_LBEN_SHIFT 4 +#define M98090_DAI_LBEN_WIDTH 1 +#define M98090_DAI_LBEN_NUM (1<<M98090_DAI_LBEN_WIDTH) +#define M98090_DAI_DMONO_MASK (1<<3) +#define M98090_DAI_DMONO_SHIFT 3 +#define M98090_DAI_DMONO_WIDTH 1 +#define M98090_DAI_DMONO_NUM (1<<M98090_DAI_DMONO_WIDTH) +#define M98090_DAI_HIZOFF_MASK (1<<2) +#define M98090_DAI_HIZOFF_SHIFT 2 +#define M98090_DAI_HIZOFF_WIDTH 1 +#define M98090_DAI_SDOEN_MASK (1<<1) +#define M98090_DAI_SDOEN_SHIFT 1 +#define M98090_DAI_SDOEN_WIDTH 1 +#define M98090_DAI_SDIEN_MASK (1<<0) +#define M98090_DAI_SDIEN_SHIFT 0 +#define M98090_DAI_SDIEN_WIDTH 1 + +/* + * M98090_REG_DAI_FILTERS + */ +#define M98090_DAI_FLT_MODE_MASK (1<<7) +#define M98090_DAI_FLT_MODE_SHIFT 7 +#define M98090_DAI_FLT_MODE_WIDTH 1 +#define M98090_DAI_FLT_AHPF_MASK (1<<6) +#define M98090_DAI_FLT_AHPF_SHIFT 6 +#define M98090_DAI_FLT_AHPF_WIDTH 1 +#define M98090_DAI_FLT_AHPF_NUM (1<<M98090_DAI_FLT_AHPF_WIDTH) +#define M98090_DAI_FLT_DHPF_MASK (1<<5) +#define M98090_DAI_FLT_DHPF_SHIFT 5 +#define M98090_DAI_FLT_DHPF_WIDTH 1 +#define M98090_DAI_FLT_DHPF_NUM (1<<M98090_DAI_FLT_DHPF_WIDTH) +#define M98090_DAI_FLT_DHF_MASK (1<<4) +#define M98090_DAI_FLT_DHF_SHIFT 4 +#define M98090_DAI_FLT_DHF_WIDTH 1 +#define M98090_DAI_FLT_DMIC34MODE_MASK (1<<3) +#define M98090_DAI_FLT_DMIC34MODE_SHIFT 3 +#define M98090_DAI_FLT_DMIC34MODE_WIDTH 1 +#define M98090_DAI_FLT_DMIC34HPF_MASK (1<<2) +#define M98090_DAI_FLT_DMIC34HPF_SHIFT 2 +#define M98090_DAI_FLT_DMIC34HPF_WIDTH 1 +#define M98090_DAI_FLT_DMIC34HPF_NUM (1<<M98090_DAI_FLT_DMIC34HPF_WIDTH) + +/* + * M98090_REG_DAI_LVL + */ +#define M98090_DAI_LVL_DVM_MASK (1<<7) +#define M98090_DAI_LVL_DVM_SHIFT 7 +#define M98090_DAI_LVL_DVM_WIDTH 1 +#define M98090_DAI_LVL_DVG_MASK (3<<4) +#define M98090_DAI_LVL_DVG_SHIFT 4 +#define M98090_DAI_LVL_DVG_WIDTH 2 +#define M98090_DAI_LVL_DVG_NUM (1<<M98090_DAI_LVL_DVG_WIDTH) +#define M98090_DAI_LVL_DV_MASK (15<<0) +#define M98090_DAI_LVL_DV_SHIFT 0 +#define M98090_DAI_LVL_DV_WIDTH 4 +#define M98090_DAI_LVL_DV_NUM (1<<M98090_DAI_LVL_DV_WIDTH) + +/* + * M98090_REG_DAI_LVL_EQ + */ +#define M98090_DAI_LVL_EQCLPN_MASK (1<<4) +#define M98090_DAI_LVL_EQCLPN_SHIFT 4 +#define M98090_DAI_LVL_EQCLPN_WIDTH 1 +#define M98090_DAI_LVL_DVEQ_MASK (15<<0) +#define M98090_DAI_LVL_DVEQ_SHIFT 0 +#define M98090_DAI_LVL_DVEQ_WIDTH 4 +#define M98090_DAI_LVL_DVEQ_NUM (1<<M98090_DAI_LVL_DVEQ_WIDTH) + +/* + * M98090_REG_MIX_HP_LEFT + */ +#define M98090_MIXHPL_MIC2_MASK (1<<5) +#define M98090_MIXHPL_MIC2_SHIFT 5 +#define M98090_MIXHPL_MIC2_WIDTH 1 +#define M98090_MIXHPL_MIC1_MASK (1<<4) +#define M98090_MIXHPL_MIC1_SHIFT 4 +#define M98090_MIXHPL_MIC1_WIDTH 1 +#define M98090_MIXHPL_LINEB_MASK (1<<3) +#define M98090_MIXHPL_LINEB_SHIFT 3 +#define M98090_MIXHPL_LINEB_WIDTH 1 +#define M98090_MIXHPL_LINEA_MASK (1<<2) +#define M98090_MIXHPL_LINEA_SHIFT 2 +#define M98090_MIXHPL_LINEA_WIDTH 1 +#define M98090_MIXHPL_DACR_MASK (1<<1) +#define M98090_MIXHPL_DACR_SHIFT 1 +#define M98090_MIXHPL_DACR_WIDTH 1 +#define M98090_MIXHPL_DACL_MASK (1<<0) +#define M98090_MIXHPL_DACL_SHIFT 0 +#define M98090_MIXHPL_DACL_WIDTH 1 +#define M98090_MIXHPL_MASK (63<<0) +#define M98090_MIXHPL_SHIFT 0 +#define M98090_MIXHPL_WIDTH 6 + +/* + * M98090_REG_MIX_HP_RIGHT + */ +#define M98090_MIXHPR_MIC2_MASK (1<<5) +#define M98090_MIXHPR_MIC2_SHIFT 5 +#define M98090_MIXHPR_MIC2_WIDTH 1 +#define M98090_MIXHPR_MIC1_MASK (1<<4) +#define M98090_MIXHPR_MIC1_SHIFT 4 +#define M98090_MIXHPR_MIC1_WIDTH 1 +#define M98090_MIXHPR_LINEB_MASK (1<<3) +#define M98090_MIXHPR_LINEB_SHIFT 3 +#define M98090_MIXHPR_LINEB_WIDTH 1 +#define M98090_MIXHPR_LINEA_MASK (1<<2) +#define M98090_MIXHPR_LINEA_SHIFT 2 +#define M98090_MIXHPR_LINEA_WIDTH 1 +#define M98090_MIXHPR_DACR_MASK (1<<1) +#define M98090_MIXHPR_DACR_SHIFT 1 +#define M98090_MIXHPR_DACR_WIDTH 1 +#define M98090_MIXHPR_DACL_MASK (1<<0) +#define M98090_MIXHPR_DACL_SHIFT 0 +#define M98090_MIXHPR_DACL_WIDTH 1 +#define M98090_MIXHPR_MASK (63<<0) +#define M98090_MIXHPR_SHIFT 0 +#define M98090_MIXHPR_WIDTH 6 + +/* + * M98090_REG_MIX_HP_CNTL + */ +#define M98090_MIXHPRSEL_MASK (1<<5) +#define M98090_MIXHPRSEL_SHIFT 5 +#define M98090_MIXHPRSEL_WIDTH 1 +#define M98090_MIXHPLSEL_MASK (1<<4) +#define M98090_MIXHPLSEL_SHIFT 4 +#define M98090_MIXHPLSEL_WIDTH 1 +#define M98090_MIXHPRG_MASK (3<<2) +#define M98090_MIXHPRG_SHIFT 2 +#define M98090_MIXHPRG_WIDTH 2 +#define M98090_MIXHPRG_NUM (1<<M98090_MIXHPRG_WIDTH) +#define M98090_MIXHPLG_MASK (3<<0) +#define M98090_MIXHPLG_SHIFT 0 +#define M98090_MIXHPLG_WIDTH 2 +#define M98090_MIXHPLG_NUM (1<<M98090_MIXHPLG_WIDTH) + +/* + * M98090_REG_LVL_HP_LEFT + */ +#define M98090_HPLM_MASK (1<<7) +#define M98090_HPLM_SHIFT 7 +#define M98090_HPLM_WIDTH 1 +#define M98090_HPVOLL_MASK (31<<0) +#define M98090_HPVOLL_SHIFT 0 +#define M98090_HPVOLL_WIDTH 5 +#define M98090_HPVOLL_NUM (1<<M98090_HPVOLL_WIDTH) + +/* + * M98090_REG_LVL_HP_RIGHT + */ +#define M98090_HPRM_MASK (1<<7) +#define M98090_HPRM_SHIFT 7 +#define M98090_HPRM_WIDTH 1 +#define M98090_HPVOLR_MASK (31<<0) +#define M98090_HPVOLR_SHIFT 0 +#define M98090_HPVOLR_WIDTH 5 +#define M98090_HPVOLR_NUM (1<<M98090_HPVOLR_WIDTH) + +/* + * M98090_REG_MIX_SPK_LEFT + */ +#define M98090_MIXSPL_MIC2_MASK (1<<5) +#define M98090_MIXSPL_MIC2_SHIFT 5 +#define M98090_MIXSPL_MIC2_WIDTH 1 +#define M98090_MIXSPL_MIC1_MASK (1<<4) +#define M98090_MIXSPL_MIC1_SHIFT 4 +#define M98090_MIXSPL_MIC1_WIDTH 1 +#define M98090_MIXSPL_LINEB_MASK (1<<3) +#define M98090_MIXSPL_LINEB_SHIFT 3 +#define M98090_MIXSPL_LINEB_WIDTH 1 +#define M98090_MIXSPL_LINEA_MASK (1<<2) +#define M98090_MIXSPL_LINEA_SHIFT 2 +#define M98090_MIXSPL_LINEA_WIDTH 1 +#define M98090_MIXSPL_DACR_MASK (1<<1) +#define M98090_MIXSPL_DACR_SHIFT 1 +#define M98090_MIXSPL_DACR_WIDTH 1 +#define M98090_MIXSPL_DACL_MASK (1<<0) +#define M98090_MIXSPL_DACL_SHIFT 0 +#define M98090_MIXSPL_DACL_WIDTH 1 +#define M98090_MIXSPL_MASK (63<<0) +#define M98090_MIXSPL_SHIFT 0 +#define M98090_MIXSPL_WIDTH 6 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 + + +/* + * M98090_REG_MIX_SPK_RIGHT + */ +#define M98090_SPK_SLAVE_MASK (1<<6) +#define M98090_SPK_SLAVE_SHIFT 6 +#define M98090_SPK_SLAVE_WIDTH 1 +#define M98090_MIXSPR_MIC2_MASK (1<<5) +#define M98090_MIXSPR_MIC2_SHIFT 5 +#define M98090_MIXSPR_MIC2_WIDTH 1 +#define M98090_MIXSPR_MIC1_MASK (1<<4) +#define M98090_MIXSPR_MIC1_SHIFT 4 +#define M98090_MIXSPR_MIC1_WIDTH 1 +#define M98090_MIXSPR_LINEB_MASK (1<<3) +#define M98090_MIXSPR_LINEB_SHIFT 3 +#define M98090_MIXSPR_LINEB_WIDTH 1 +#define M98090_MIXSPR_LINEA_MASK (1<<2) +#define M98090_MIXSPR_LINEA_SHIFT 2 +#define M98090_MIXSPR_LINEA_WIDTH 1 +#define M98090_MIXSPR_DACR_MASK (1<<1) +#define M98090_MIXSPR_DACR_SHIFT 1 +#define M98090_MIXSPR_DACR_WIDTH 1 +#define M98090_MIXSPR_DACL_MASK (1<<0) +#define M98090_MIXSPR_DACL_SHIFT 0 +#define M98090_MIXSPR_DACL_WIDTH 1 +#define M98090_MIXSPR_MASK (63<<0) +#define M98090_MIXSPR_SHIFT 0 +#define M98090_MIXSPR_WIDTH 6 + +/* + * M98090_REG_MIX_SPK_CNTL + */ +#define M98090_MIXSPRG_MASK (3<<2) +#define M98090_MIXSPRG_SHIFT 2 +#define M98090_MIXSPRG_WIDTH 2 +#define M98090_MIXSPRG_NUM (1<<M98090_MIXSPRG_WIDTH) +#define M98090_MIXSPLG_MASK (3<<0) +#define M98090_MIXSPLG_SHIFT 0 +#define M98090_MIXSPLG_WIDTH 2 +#define M98090_MIXSPLG_NUM (1<<M98090_MIXSPLG_WIDTH) + +/* + * M98090_REG_LVL_SPK_LEFT + */ +#define M98090_SPLM_MASK (1<<7) +#define M98090_SPLM_SHIFT 7 +#define M98090_SPLM_WIDTH 1 +#define M98090_SPVOLL_MASK (63<<0) +#define M98090_SPVOLL_SHIFT 0 +#define M98090_SPVOLL_WIDTH 6 +#define M98090_SPVOLL_NUM 40 + +/* + * M98090_REG_LVL_SPK_RIGHT + */ +#define M98090_SPRM_MASK (1<<7) +#define M98090_SPRM_SHIFT 7 +#define M98090_SPRM_WIDTH 1 +#define M98090_SPVOLR_MASK (63<<0) +#define M98090_SPVOLR_SHIFT 0 +#define M98090_SPVOLR_WIDTH 6 +#define M98090_SPVOLR_NUM 40 + +/* + * M98090_REG_ALC_TIMING + */ +#define M98090_ALCEN_MASK (1<<7) +#define M98090_ALCEN_SHIFT 7 +#define M98090_ALCEN_WIDTH 1 +#define M98090_ALCEN_NUM (1<<M98090_ALCEN_WIDTH) +#define M98090_ALCRLS_MASK (7<<4) +#define M98090_ALCRLS_SHIFT 4 +#define M98090_ALCRLS_WIDTH 3 +#define M98090_ALCATK_MASK (7<<0) +#define M98090_ALCATK_SHIFT 0 +#define M98090_ALCATK_WIDTH 3 + +/* + * M98090_REG_ALC_CMPR + */ +#define M98090_ALCCMP_MASK (7<<5) +#define M98090_ALCCMP_SHIFT 5 +#define M98090_ALCCMP_WIDTH 3 +#define M98090_ALCTHC_MASK (31<<0) +#define M98090_ALCTHC_SHIFT 0 +#define M98090_ALCTHC_WIDTH 5 +#define M98090_ALCTHC_NUM (1<<M98090_ALCTHC_WIDTH) + +/* + * M98090_REG_ALC_EXP + */ +#define M98090_ALCEXP_MASK (7<<5) +#define M98090_ALCEXP_SHIFT 5 +#define M98090_ALCEXP_WIDTH 3 +#define M98090_ALCTHE_MASK (31<<0) +#define M98090_ALCTHE_SHIFT 0 +#define M98090_ALCTHE_WIDTH 5 +#define M98090_ALCTHE_NUM (1<<M98090_ALCTHE_WIDTH) + +/* + * M98090_REG_LVL_ALC + */ +#define M98090_ALCG_MASK (31<<0) +#define M98090_ALCG_SHIFT 0 +#define M98090_ALCG_WIDTH 5 +#define M98090_ALCG_NUM 13 + +/* + * M98090_REG_MIX_RCV_LEFT + */ +#define M98090_MIXRCVL_MIC2_MASK (1<<5) +#define M98090_MIXRCVL_MIC2_SHIFT 5 +#define M98090_MIXRCVL_MIC2_WIDTH 1 +#define M98090_MIXRCVL_MIC1_MASK (1<<4) +#define M98090_MIXRCVL_MIC1_SHIFT 4 +#define M98090_MIXRCVL_MIC1_WIDTH 1 +#define M98090_MIXRCVL_LINEB_MASK (1<<3) +#define M98090_MIXRCVL_LINEB_SHIFT 3 +#define M98090_MIXRCVL_LINEB_WIDTH 1 +#define M98090_MIXRCVL_LINEA_MASK (1<<2) +#define M98090_MIXRCVL_LINEA_SHIFT 2 +#define M98090_MIXRCVL_LINEA_WIDTH 1 +#define M98090_MIXRCVL_DACR_MASK (1<<1) +#define M98090_MIXRCVL_DACR_SHIFT 1 +#define M98090_MIXRCVL_DACR_WIDTH 1 +#define M98090_MIXRCVL_DACL_MASK (1<<0) +#define M98090_MIXRCVL_DACL_SHIFT 0 +#define M98090_MIXRCVL_DACL_WIDTH 1 +#define M98090_MIXRCVL_MASK (63<<0) +#define M98090_MIXRCVL_SHIFT 0 +#define M98090_MIXRCVL_WIDTH 6 + +/* + * M98090_REG_MIX_RCV_CTRL_LEFT + */ +#define M98090_MIXRCVLG_MASK (3<<0) +#define M98090_MIXRCVLG_SHIFT 0 +#define M98090_MIXRCVLG_WIDTH 2 +#define M98090_MIXRCVLG_NUM (1<<M98090_MIXRCVLG_WIDTH) + +/* + * M98090_REG_LVL_RCV_LEFT + */ +#define M98090_RCVLM_MASK (1<<7) +#define M98090_RCVLM_SHIFT 7 +#define M98090_RCVLM_WIDTH 1 +#define M98090_RCVLVOL_MASK (31<<0) +#define M98090_RCVLVOL_SHIFT 0 +#define M98090_RCVLVOL_WIDTH 5 +#define M98090_RCVLVOL_NUM (1<<M98090_RCVLVOL_WIDTH) + +/* + * M98090_REG_MIX_RCV_RIGHT + */ +#define M98090_LINMOD_MASK (1<<7) +#define M98090_LINMOD_SHIFT 7 +#define M98090_LINMOD_WIDTH 1 +#define M98090_MIXRCVR_MIC2_MASK (1<<5) +#define M98090_MIXRCVR_MIC2_SHIFT 5 +#define M98090_MIXRCVR_MIC2_WIDTH 1 +#define M98090_MIXRCVR_MIC1_MASK (1<<4) +#define M98090_MIXRCVR_MIC1_SHIFT 4 +#define M98090_MIXRCVR_MIC1_WIDTH 1 +#define M98090_MIXRCVR_LINEB_MASK (1<<3) +#define M98090_MIXRCVR_LINEB_SHIFT 3 +#define M98090_MIXRCVR_LINEB_WIDTH 1 +#define M98090_MIXRCVR_LINEA_MASK (1<<2) +#define M98090_MIXRCVR_LINEA_SHIFT 2 +#define M98090_MIXRCVR_LINEA_WIDTH 1 +#define M98090_MIXRCVR_DACR_MASK (1<<1) +#define M98090_MIXRCVR_DACR_SHIFT 1 +#define M98090_MIXRCVR_DACR_WIDTH 1 +#define M98090_MIXRCVR_DACL_MASK (1<<0) +#define M98090_MIXRCVR_DACL_SHIFT 0 +#define M98090_MIXRCVR_DACL_WIDTH 1 +#define M98090_MIXRCVR_MASK (63<<0) +#define M98090_MIXRCVR_SHIFT 0 +#define M98090_MIXRCVR_WIDTH 6 + +/* + * M98090_REG_MIX_RCV_CNTL_RIGHT + */ +#define M98090_MIXRCVRG_MASK (3<<0) +#define M98090_MIXRCVRG_SHIFT 0 +#define M98090_MIXRCVRG_WIDTH 2 +#define M98090_MIXRCVRG_NUM (1<<M98090_MIXRCVRG_WIDTH) + +/* + * M98090_REG_LVL_RCV_RIGHT + */ +#define M98090_RCVRM_MASK (1<<7) +#define M98090_RCVRM_SHIFT 7 +#define M98090_RCVRM_WIDTH 1 +#define M98090_RCVRVOL_MASK (31<<0) +#define M98090_RCVRVOL_SHIFT 0 +#define M98090_RCVRVOL_WIDTH 5 +#define M98090_RCVRVOL_NUM (1<<M98090_RCVRVOL_WIDTH) + +/* + * M98090_REG_CFG_JACK + */ +#define M98090_JDETEN_MASK (1<<7) +#define M98090_JDETEN_SHIFT 7 +#define M98090_JDETEN_WIDTH 1 +#define M98090_JDWK_MASK (1<<6) +#define M98090_JDWK_SHIFT 6 +#define M98090_JDWK_WIDTH 1 +#define M98090_JDEB_MASK (3<<0) +#define M98090_JDEB_SHIFT 0 +#define M98090_JDEB_WIDTH 2 +#define M98090_JDEB_25MS (0<<0) +#define M98090_JDEB_50MS (1<<0) +#define M98090_JDEB_100MS (2<<0) +#define M98090_JDEB_200MS (3<<0) + +/* + * M98090_REG_PWR_EN_IN + */ +#define M98090_PWR_MBEN_MASK (1<<4) +#define M98090_PWR_MBEN_SHIFT 4 +#define M98090_PWR_MBEN_WIDTH 1 +#define M98090_PWR_LINEAEN_MASK (1<<3) +#define M98090_PWR_LINEAEN_SHIFT 3 +#define M98090_PWR_LINEAEN_WIDTH 1 +#define M98090_PWR_LINEBEN_MASK (1<<2) +#define M98090_PWR_LINEBEN_SHIFT 2 +#define M98090_PWR_LINEBEN_WIDTH 1 +#define M98090_PWR_ADREN_MASK (1<<1) +#define M98090_PWR_ADREN_SHIFT 1 +#define M98090_PWR_ADREN_WIDTH 1 +#define M98090_PWR_ADLEN_MASK (1<<0) +#define M98090_PWR_ADLEN_SHIFT 0 +#define M98090_PWR_ADLEN_WIDTH 1 + +/* + * M98090_REG_PWR_EN_OUT + */ +#define M98090_PWR_HPREN_MASK (1<<7) +#define M98090_PWR_HPREN_SHIFT 7 +#define M98090_PWR_HPREN_WIDTH 1 +#define M98090_PWR_HPLEN_MASK (1<<6) +#define M98090_PWR_HPLEN_SHIFT 6 +#define M98090_PWR_HPLEN_WIDTH 1 +#define M98090_PWR_SPREN_MASK (1<<5) +#define M98090_PWR_SPREN_SHIFT 5 +#define M98090_PWR_SPREN_WIDTH 1 +#define M98090_PWR_SPLEN_MASK (1<<4) +#define M98090_PWR_SPLEN_SHIFT 4 +#define M98090_PWR_SPLEN_WIDTH 1 +#define M98090_PWR_RCVLEN_MASK (1<<3) +#define M98090_PWR_RCVLEN_SHIFT 3 +#define M98090_PWR_RCVLEN_WIDTH 1 +#define M98090_PWR_RCVREN_MASK (1<<2) +#define M98090_PWR_RCVREN_SHIFT 2 +#define M98090_PWR_RCVREN_WIDTH 1 +#define M98090_PWR_DAREN_MASK (1<<1) +#define M98090_PWR_DAREN_SHIFT 1 +#define M98090_PWR_DAREN_WIDTH 1 +#define M98090_PWR_DALEN_MASK (1<<0) +#define M98090_PWR_DALEN_SHIFT 0 +#define M98090_PWR_DALEN_WIDTH 1 + +/* + * M98090_REG_CFG_LVL + */ +#define M98090_ZDENN_MASK (1<<2) +#define M98090_ZDENN_SHIFT 2 +#define M98090_ZDENN_WIDTH 1 +#define M98090_VS2ENN_MASK (1<<1) +#define M98090_VS2ENN_SHIFT 1 +#define M98090_VS2ENN_WIDTH 1 +#define M98090_VSENN_MASK (1<<0) +#define M98090_VSENN_SHIFT 0 +#define M98090_VSENN_WIDTH 1 + +/* + * M98090_REG_DSP_EQ_EN + */ +#define M98090_DMIC34BQEN_MASK (1<<4) +#define M98090_DMIC34BQEN_SHIFT 4 +#define M98090_DMIC34BQEN_WIDTH 1 +#define M98090_DMIC34BQEN_NUM (1<<M98090_DMIC34BQEN_WIDTH) +#define M98090_ADCBQEN_MASK (1<<3) +#define M98090_ADCBQEN_SHIFT 3 +#define M98090_ADCBQEN_WIDTH 1 +#define M98090_ADCBQEN_NUM (1<<M98090_ADCBQEN_WIDTH) +#define M98090_EQ3BANDEN_MASK (1<<2) +#define M98090_EQ3BANDEN_SHIFT 2 +#define M98090_EQ3BANDEN_WIDTH 1 +#define M98090_EQ3BANDEN_NUM (1<<M98090_EQ3BANDEN_WIDTH) +#define M98090_EQ5BANDEN_MASK (1<<1) +#define M98090_EQ5BANDEN_SHIFT 1 +#define M98090_EQ5BANDEN_WIDTH 1 +#define M98090_EQ5BANDEN_NUM (1<<M98090_EQ5BANDEN_WIDTH) +#define M98090_EQ7BANDEN_MASK (1<<0) +#define M98090_EQ7BANDEN_SHIFT 0 +#define M98090_EQ7BANDEN_WIDTH 1 +#define M98090_EQ7BANDEN_NUM (1<<M98090_EQ7BANDEN_WIDTH) + +/* + * M98090_REG_BIAS_CNTL + */ +#define M98090_VCM_MODE_MASK (1<<0) +#define M98090_VCM_MODE_SHIFT 0 +#define M98090_VCM_MODE_WIDTH 1 +#define M98090_VCM_MODE_NUM (1<<M98090_VCM_MODE_WIDTH) + +/* + * M98090_REG_DAC_CFG + */ +#define M98090_DAC_PERFMODE_MASK (1<<1) +#define M98090_DAC_PERFMODE_SHIFT 1 +#define M98090_DAC_PERFMODE_WIDTH 1 +#define M98090_DAC_PERFMODE_NUM (1<<M98090_DAC_PERFMODE_WIDTH) +#define M98090_DACHP_MASK (1<<0) +#define M98090_DACHP_SHIFT 0 +#define M98090_DACHP_WIDTH 1 +#define M98090_DACHP_NUM (1<<M98090_DACHP_WIDTH) + +/* + * M98090_REG_ADC_CFG + */ +#define M98090_ADC_OSR128_MASK (1<<2) +#define M98090_ADC_OSR128_SHIFT 2 +#define M98090_ADC_OSR128_WIDTH 1 +#define M98090_ADCDITHER_MASK (1<<1) +#define M98090_ADCDITHER_SHIFT 1 +#define M98090_ADCDITHER_WIDTH 1 +#define M98090_ADCDITHER_NUM (1<<M98090_ADCDITHER_WIDTH) +#define M98090_ADCHP_MASK (1<<0) +#define M98090_ADCHP_SHIFT 0 +#define M98090_ADCHP_WIDTH 1 +#define M98090_ADCHP_NUM (1<<M98090_ADCHP_WIDTH) + +/* + * M98090_REG_PWR_SYS + */ +#define M98090_PWR_SHDNN_MASK (1<<7) +#define M98090_PWR_SHDNN_SHIFT 7 +#define M98090_PWR_SHDNN_WIDTH 1 + +/* + * M98090_REG_EQ_BASE + */ +#define M98090_B0_1_HI_MASK (255<<0) +#define M98090_B0_1_HI_SHIFT 0 +#define M98090_B0_1_HI_WIDTH 8 +#define M98090_B0_1_MID_MASK (255<<0) +#define M98090_B0_1_MID_SHIFT 0 +#define M98090_B0_1_MID_WIDTH 8 +#define M98090_B0_1_LO_MASK (255<<0) +#define M98090_B0_1_LO_SHIFT 0 +#define M98090_B0_1_LO_WIDTH 8 +#define M98090_B1_1_HI_MASK (255<<0) +#define M98090_B1_1_HI_SHIFT 0 +#define M98090_B1_1_HI_WIDTH 8 +#define M98090_B1_1_MID_MASK (255<<0) +#define M98090_B1_1_MID_SHIFT 0 +#define M98090_B1_1_MID_WIDTH 8 +#define M98090_B1_1_LO_MASK (255<<0) +#define M98090_B1_1_LO_SHIFT 0 +#define M98090_B1_1_LO_WIDTH 8 +#define M98090_B2_1_HI_MASK (255<<0) +#define M98090_B2_1_HI_SHIFT 0 +#define M98090_B2_1_HI_WIDTH 8 +#define M98090_B2_1_MID_MASK (255<<0) +#define M98090_B2_1_MID_SHIFT 0 +#define M98090_B2_1_MID_WIDTH 8 +#define M98090_B2_1_LO_MASK (255<<0) +#define M98090_B2_1_LO_SHIFT 0 +#define M98090_B2_1_LO_WIDTH 8 +#define M98090_A1_1_HI_MASK (255<<0) +#define M98090_A1_1_HI_SHIFT 0 +#define M98090_A1_1_HI_WIDTH 8 +#define M98090_A1_1_MID_MASK (255<<0) +#define M98090_A1_1_MID_SHIFT 0 +#define M98090_A1_1_MID_WIDTH 8 +#define M98090_A1_1_LO_MASK (255<<0) +#define M98090_A1_1_LO_SHIFT 0 +#define M98090_A1_1_LO_WIDTH 8 +#define M98090_A2_1_HI_MASK (255<<0) +#define M98090_A2_1_HI_SHIFT 0 +#define M98090_A2_1_HI_WIDTH 8 +#define M98090_A2_1_MID_MASK (255<<0) +#define M98090_A2_1_MID_SHIFT 0 +#define M98090_A2_1_MID_WIDTH 8 +#define M98090_A2_1_LO_MASK (255<<0) +#define M98090_A2_1_LO_SHIFT 0 +#define M98090_A2_1_LO_WIDTH 8 + +#define M98090_COEFS_PER_BAND 5 +#define M98090_COEFS_BLK_SZ (M98090_COEFS_PER_BAND * 3) +#define M98090_COEFS_MAX_SZ (M98090_COEFS_BLK_SZ * 7) + +/* + * M98090_REG_BIQUAD_BASE + */ +#define M98090_ADC_B0_HI_MASK (255<<0) +#define M98090_ADC_B0_HI_SHIFT 0 +#define M98090_ADC_B0_HI_WIDTH 8 +#define M98090_ADC_B0_MID_MASK (255<<0) +#define M98090_ADC_B0_MID_SHIFT 0 +#define M98090_ADC_B0_MID_WIDTH 8 +#define M98090_ADC_B0_LO_MASK (255<<0) +#define M98090_ADC_B0_LO_SHIFT 0 +#define M98090_ADC_B0_LO_WIDTH 8 +#define M98090_ADC_B1_HI_MASK (255<<0) +#define M98090_ADC_B1_HI_SHIFT 0 +#define M98090_ADC_B1_HI_WIDTH 8 +#define M98090_ADC_B1_MID_MASK (255<<0) +#define M98090_ADC_B1_MID_SHIFT 0 +#define M98090_ADC_B1_MID_WIDTH 8 +#define M98090_ADC_B1_LO_MASK (255<<0) +#define M98090_ADC_B1_LO_SHIFT 0 +#define M98090_ADC_B1_LO_WIDTH 8 +#define M98090_ADC_B2_HI_MASK (255<<0) +#define M98090_ADC_B2_HI_SHIFT 0 +#define M98090_ADC_B2_HI_WIDTH 8 +#define M98090_ADC_B2_MID_MASK (255<<0) +#define M98090_ADC_B2_MID_SHIFT 0 +#define M98090_ADC_B2_MID_WIDTH 8 +#define M98090_ADC_B2_LO_MASK (255<<0) +#define M98090_ADC_B2_LO_SHIFT 0 +#define M98090_ADC_B2_LO_WIDTH 8 +#define M98090_ADC_A1_HI_MASK (255<<0) +#define M98090_ADC_A1_HI_SHIFT 0 +#define M98090_ADC_A1_HI_WIDTH 8 +#define M98090_ADC_A1_MID_MASK (255<<0) +#define M98090_ADC_A1_MID_SHIFT 0 +#define M98090_ADC_A1_MID_WIDTH 8 +#define M98090_ADC_A1_LO_MASK (255<<0) +#define M98090_ADC_A1_LO_SHIFT 0 +#define M98090_ADC_A1_LO_WIDTH 8 +#define M98090_ADC_A2_HI_MASK (255<<0) +#define M98090_ADC_A2_HI_SHIFT 0 +#define M98090_ADC_A2_HI_WIDTH 8 +#define M98090_ADC_A2_MID_MASK (255<<0) +#define M98090_ADC_A2_MID_SHIFT 0 +#define M98090_ADC_A2_MID_WIDTH 8 +#define M98090_ADC_A2_LO_MASK (255<<0) +#define M98090_ADC_A2_LO_SHIFT 0 +#define M98090_ADC_A2_LO_WIDTH 8 + +/* + * M98090_REG_DMIC3_VOLUME + */ +#define M98090_DMIC_AV3G_MASK (7<<4) +#define M98090_DMIC_AV3G_SHIFT 4 +#define M98090_DMIC_AV3G_WIDTH 3 +#define M98090_DMIC_AV3G_NUM (1<<M98090_DMIC_AV3G_WIDTH) +#define M98090_DMIC_AV3_MASK (15<<0) +#define M98090_DMIC_AV3_SHIFT 0 +#define M98090_DMIC_AV3_WIDTH 4 +#define M98090_DMIC_AV3_NUM (1<<M98090_DMIC_AV3_WIDTH) + +/* + * M98090_REG_DMIC4_VOLUME + */ +#define M98090_DMIC_AV4G_MASK (7<<4) +#define M98090_DMIC_AV4G_SHIFT 4 +#define M98090_DMIC_AV4G_WIDTH 3 +#define M98090_DMIC_AV4G_NUM (1<<M98090_DMIC_AV4G_WIDTH) +#define M98090_DMIC_AV4_MASK (15<<0) +#define M98090_DMIC_AV4_SHIFT 0 +#define M98090_DMIC_AV4_WIDTH 4 +#define M98090_DMIC_AV4_NUM (1<<M98090_DMIC_AV4_WIDTH) + +/* + * M98090_REG_DMIC34_BQ_PREATTEN + */ +#define M98090_AV34BQ_MASK (15<<0) +#define M98090_AV34BQ_SHIFT 0 +#define M98090_AV34BQ_WIDTH 4 +#define M98090_AV34BQ_NUM (1<<M98090_AV34BQ_WIDTH) + +/* + * M98090_REG_RECORD_TDM_SLOT + */ +#define M98090_DAI_TDM_SLOTADCL_MASK (3<<6) +#define M98090_DAI_TDM_SLOTADCL_SHIFT 6 +#define M98090_DAI_TDM_SLOTADCL_WIDTH 2 +#define M98090_DAI_TDM_SLOTADCL_NUM (1<<M98090_DAI_TDM_SLOTADCL_WIDTH) +#define M98090_DAI_TDM_SLOTADCR_MASK (3<<4) +#define M98090_DAI_TDM_SLOTADCR_SHIFT 4 +#define M98090_DAI_TDM_SLOTADCR_WIDTH 2 +#define M98090_DAI_TDM_SLOTADCR_NUM (1<<M98090_DAI_TDM_SLOTADCR_WIDTH) +#define M98090_DAI_TDM_SLOTDMIC3_MASK (3<<2) +#define M98090_DAI_TDM_SLOTDMIC3_SHIFT 2 +#define M98090_DAI_TDM_SLOTDMIC3_WIDTH 2 +#define M98090_DAI_TDM_SLOTDMIC3_NUM (1<<M98090_DAI_TDM_SLOTDMIC3_WIDTH) +#define M98090_DAI_TDM_SLOTDMIC4_MASK (3<<0) +#define M98090_DAI_TDM_SLOTDMIC4_SHIFT 0 +#define M98090_DAI_TDM_SLOTDMIC4_WIDTH 2 +#define M98090_DAI_TDM_SLOTDMIC4_NUM (1<<M98090_DAI_TDM_SLOTDMIC4_WIDTH) + +/* + * M98090_REG_SAMPLE_RATE + */ +#define M98090_DMIC34_ZEROPAD_MASK (1<<4) +#define M98090_DMIC34_ZEROPAD_SHIFT 4 +#define M98090_DMIC34_ZEROPAD_WIDTH 1 +#define M98090_DMIC34_ZEROPAD_NUM (1<<M98090_DIGMIC4_WIDTH) +#define M98090_DMIC34_SRDIV_MASK (7<<0) +#define M98090_DMIC34_SRDIV_SHIFT 0 +#define M98090_DMIC34_SRDIV_WIDTH 3 + +/* + * M98090_REG_DMIC34_BIQUAD_BASE + */ +#define M98090_DMIC34_B0_HI_MASK (255<<0) +#define M98090_DMIC34_B0_HI_SHIFT 0 +#define M98090_DMIC34_B0_HI_WIDTH 8 +#define M98090_DMIC34_B0_MID_MASK (255<<0) +#define M98090_DMIC34_B0_MID_SHIFT 0 +#define M98090_DMIC34_B0_MID_WIDTH 8 +#define M98090_DMIC34_B0_LO_MASK (255<<0) +#define M98090_DMIC34_B0_LO_SHIFT 0 +#define M98090_DMIC34_B0_LO_WIDTH 8 +#define M98090_DMIC34_B1_HI_MASK (255<<0) +#define M98090_DMIC34_B1_HI_SHIFT 0 +#define M98090_DMIC34_B1_HI_WIDTH 8 +#define M98090_DMIC34_B1_MID_MASK (255<<0) +#define M98090_DMIC34_B1_MID_SHIFT 0 +#define M98090_DMIC34_B1_MID_WIDTH 8 +#define M98090_DMIC34_B1_LO_MASK (255<<0) +#define M98090_DMIC34_B1_LO_SHIFT 0 +#define M98090_DMIC34_B1_LO_WIDTH 8 +#define M98090_DMIC34_B2_HI_MASK (255<<0) +#define M98090_DMIC34_B2_HI_SHIFT 0 +#define M98090_DMIC34_B2_HI_WIDTH 8 +#define M98090_DMIC34_B2_MID_MASK (255<<0) +#define M98090_DMIC34_B2_MID_SHIFT 0 +#define M98090_DMIC34_B2_MID_WIDTH 8 +#define M98090_DMIC34_B2_LO_MASK (255<<0) +#define M98090_DMIC34_B2_LO_SHIFT 0 +#define M98090_DMIC34_B2_LO_WIDTH 8 +#define M98090_DMIC34_A1_HI_MASK (255<<0) +#define M98090_DMIC34_A1_HI_SHIFT 0 +#define M98090_DMIC34_A1_HI_WIDTH 8 +#define M98090_DMIC34_A1_MID_MASK (255<<0) +#define M98090_DMIC34_A1_MID_SHIFT 0 +#define M98090_DMIC34_A1_MID_WIDTH 8 +#define M98090_DMIC34_A1_LO_MASK (255<<0) +#define M98090_DMIC34_A1_LO_SHIFT 0 +#define M98090_DMIC34_A1_LO_WIDTH 8 +#define M98090_DMIC34_A2_HI_MASK (255<<0) +#define M98090_DMIC34_A2_HI_SHIFT 0 +#define M98090_DMIC34_A2_HI_WIDTH 8 +#define M98090_DMIC34_A2_MID_MASK (255<<0) +#define M98090_DMIC34_A2_MID_SHIFT 0 +#define M98090_DMIC34_A2_MID_WIDTH 8 +#define M98090_DMIC34_A2_LO_MASK (255<<0) +#define M98090_DMIC34_A2_LO_SHIFT 0 +#define M98090_DMIC34_A2_LO_WIDTH 8 + +#define M98090_JACK_STATE_NO_HEADSET 0 +#define M98090_JACK_STATE_NO_HEADSET_2 1 +#define M98090_JACK_STATE_HEADPHONE 2 +#define M98090_JACK_STATE_HEADSET 3 + +/* + * M98090_REG_REV_ID + */ +#define M98090_REVID_MASK (255<<0) +#define M98090_REVID_SHIFT 0 +#define M98090_REVID_WIDTH 8 +#define M98090_REVID_NUM (1<<M98090_REVID_WIDTH) + +#define M98090_BYTE1(w) ((w >> 8) & 0xff) +#define M98090_BYTE0(w) (w & 0xff) + +/* Silicon revision number */ +#define M98090_REVA 0x40 +#define M98091_REVA 0x50 + +enum max98090_type { + MAX98090, + MAX98091, +}; + +struct max98090_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98090_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + enum max98090_type devtype; + void *control_data; + struct max98090_pdata *pdata; + unsigned int sysclk; + unsigned int bclk; + unsigned int lrclk; + struct max98090_cdata dai[1]; + int irq; + int jack_state; + struct delayed_work jack_work; + struct snd_soc_jack *jack; + unsigned int dai_fmt; + int tdm_slots; + int tdm_width; + u8 lin_state; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mux; + unsigned int sidetone; +}; + +int max98090_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 1.7.0.4

2 2

Re: [alsa-devel] HD-audio generic parser merged to linux-next
by Takashi Iwai 23 Jan '13

23 Jan '13

At Wed, 23 Jan 2013 15:01:43 +0100, Manolo Díaz wrote: > > El mié, 23 ene 2013 a las 14:54 horas > Takashi Iwai escribió: > > >At Wed, 23 Jan 2013 14:49:32 +0100, > >Manolo Díaz wrote: > >> > >> El mié, 23 ene 2013 a las 9:30 horas > >> Takashi Iwai escribió: > >> > >> >Hi, > >> > > >> >as most of feedbacks about the new generic parser migration were > >> >positive, I decided to take a risk: now all commit that have been in > >> >test/hda-migrate branch in sound-unstable git tree are merged to > >> >for-next branch of sound git tree for better test coverage. > >> >It means that all changes are merged to linux-next, i.e. will be > >> >included in 3.9 kernel. > >> > > >> >For the early adapter people, I'm going to keep test/hda-migrate > >> >branch of sound-unstable git tree updated for a while, but the branch > >> >will be abandoned some time later. > >> > > >> >If you encounter any regression by the recent update, please let me > >> >know. > >> > > >> > > >> >thanks, > >> > > >> >Takashi > >> >_______________________________________________ > >> >Alsa-devel mailing list > >> >Alsa-devel(a)alsa-project.org > >> >http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >> > >> Hi, > >> > >> Recording and (multi stream) playing seem to work fine, but auto-mute > >> doesn't. For example, front speakers may remain muted after > >> disconnecting the HP. > >> > >> cat /sys/kernel/debug/tracing/trace > >> # tracer: nop > >> # > >> # entries-in-buffer/entries-written: 0/0 #P:2 > >> # > >> # _-----=> irqs-off > >> # / _----=> need-resched > >> # | / _---=> hardirq/softirq > >> # || / _--=> preempt-depth > >> # ||| / delay > >> # TASK-PID CPU# |||| TIMESTAMP FUNCTION > >> # | | | |||| | | > >> > >> I've connected and disconnected the HP jack several times before. > > > >If nothing appears in the tracing (suppose you enabled the events > >properly via /sys/kernel/debug/tracing/events/hda/enable), it's rather > >a hardware issue. > > Sorry, I've forgotten this step. > > >Did the auto-mute work reliably before the code change? > > > > > >Takashi > > Yes, it did. Did you mean when it worked? Was it before moving to the generic parser, or after it but before upgrading to the latest tree? In the latter case, try git bisection to figure out the culprit. But, it's anyway better to update the git branch. You are testing the too old version (there have been already 75 commits after your testing one). > New try, this time with hda events enabled. > > cat /sys/kernel/debug/tracing/trace > # tracer: nop > # > # entries-in-buffer/entries-written: 871/871 #P:2 > # > # _-----=> irqs-off > # / _----=> need-resched > # | / _---=> hardirq/softirq > # || / _--=> preempt-depth > # ||| / delay > # TASK-PID CPU# |||| TIMESTAMP FUNCTION > # | | | |||| | | > <idle>-0 [001] d.h. 4509.450172: hda_unsol_event: [0] res=4000000, res_ex=10 > <idle>-0 [001] d.h. 4513.991711: hda_unsol_event: [0] res=4000000, res_ex=10 OK, so the unsol event is generated properly. Is the alsa-info.sh output you attached previously at the state after unplugging? If so, I see no problem in the code register. The path 0x04 -> 0x29 -> 0x12 looks fine, all unmuted, power D0 and the pin set properly. If something doesn't work as expected, we need two alsa-info outputs for comparison: the working one and non-working one. Takashi

2 10

[alsa-devel] [PATCH] ASoC: wm2000: Lock state machine updates
by Mark Brown 23 Jan '13

23 Jan '13

Need to ensure we don't get confused by simultaneous updates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 31 ++++++++++++++++++++++++++++--- 1 file changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 655f43e..4d809ab 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -77,6 +77,8 @@ struct wm2000_priv { int anc_download_size; char *anc_download; + + struct mutex lock; }; static int wm2000_write(struct i2c_client *i2c, unsigned int reg, @@ -614,13 +616,20 @@ static int wm2000_anc_mode_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int anc_active = ucontrol->value.enumerated.item[0]; + int ret; if (anc_active > 1) return -EINVAL; + mutex_lock(&wm2000->lock); + wm2000->anc_active = anc_active; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static int wm2000_speaker_get(struct snd_kcontrol *kcontrol, @@ -640,13 +649,20 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); int val = ucontrol->value.enumerated.item[0]; + int ret; if (val > 1) return -EINVAL; + mutex_lock(&wm2000->lock); + wm2000->spk_ena = val; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static const struct snd_kcontrol_new wm2000_controls[] = { @@ -663,6 +679,9 @@ static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + int ret; + + mutex_lock(&wm2000->lock); if (SND_SOC_DAPM_EVENT_ON(event)) wm2000->anc_eng_ena = 1; @@ -670,7 +689,11 @@ static int wm2000_anc_power_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_OFF(event)) wm2000->anc_eng_ena = 0; - return wm2000_anc_set_mode(wm2000); + ret = wm2000_anc_set_mode(wm2000); + + mutex_unlock(&wm2000->lock); + + return ret; } static const struct snd_soc_dapm_widget wm2000_dapm_widgets[] = { @@ -800,6 +823,8 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } + mutex_init(&wm2000->lock); + dev_set_drvdata(&i2c->dev, wm2000); wm2000->regmap = devm_regmap_init_i2c(i2c, &wm2000_regmap); -- 1.7.10.4

1 0

[alsa-devel] repeated hda_codec_cleanup_stream
by Raj Upadhyaya 23 Jan '13

23 Jan '13

(Bugzilla bug title..) [Fwd: [Bug 894259] Sound is garbled on playback of OGG, MP3s, WAV files and videos with sound.] I am trying to debug why I have some sound problems with my set up. I have recently recompiled the kernel to 3.8.0-rc4. Here is a suggestion that I got recently from the debugging session. The message -> ALSA hda_codec.c:1499 hda_codec_cleanup_stream: is coming up in dmesg quite often. Is this OK, given my kernel compile options? - Raj -------- Forwarded Message -------- From: bugzilla(a)redhat.com To: raj(a)upadhyaya.com Subject: [Bug 894259] Sound is garbled on playback of OGG, MP3s, WAV files and videos with sound. Date: Wed, 23 Jan 2013 06:04:20 +0000 Product: Fedora https://bugzilla.redhat.com/show_bug.cgi?id=894259 --- Comment #17 from Raymond <superquad.vortex2(a)gmail.com> --- i think you better report the repeated hda_codec_cleanup_stream to alsa development mailing list .374225] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.374575] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.374581] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.374585] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.374588] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.374592] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.374916] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.374921] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.374925] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.374929] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.374932] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.375602] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.375607] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.375611] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.375615] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.375619] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 [ 32.376435] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x2 [ 32.376440] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x3 [ 32.376444] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x4 [ 32.376448] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x5 [ 32.376452] ALSA hda_codec.c:1499 hda_codec_cleanup_stream: NID=0x25 -- You are receiving this mail because: You reported the bug.

2 1

[alsa-devel] sound/pci/hda/patch_analog.c:4987 patch_ad1882() error: potentially derefencing uninitialized 'spec'.
by Dan Carpenter 22 Jan '13

22 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 96349b211586f3a6f2750bc4ea195bd9739465b6 commit: e87b24f711c797f1277188dc85b866bec17704a0 ALSA: hda - Rearrange for dropping static quirk codes in AD codec driver New smatch warnings: sound/pci/hda/patch_analog.c:4987 patch_ad1882() error: potentially derefencing uninitialized 'spec'. git remote add sound-unstable git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git git remote update sound-unstable git checkout e87b24f711c797f1277188dc85b866bec17704a0 vim +/spec +4987 sound/pci/hda/patch_analog.c e87b24f7 Takashi Iwai 2013-01-22 4971 #ifdef ENABLE_AD_STATIC_QUIRKS 0ac8551e Takashi Iwai 2007-06-20 4972 static int patch_ad1882(struct hda_codec *codec) 0ac8551e Takashi Iwai 2007-06-20 4973 { 0ac8551e Takashi Iwai 2007-06-20 4974 struct ad198x_spec *spec; c5a4bcd0 Takashi Iwai 2009-02-06 4975 int err, board_config; 0ac8551e Takashi Iwai 2007-06-20 4976 78bb3cb0 Takashi Iwai 2012-12-21 4977 board_config = snd_hda_check_board_config(codec, AD1882_MODELS, 78bb3cb0 Takashi Iwai 2012-12-21 4978 ad1882_models, NULL); e87b24f7 Takashi Iwai 2013-01-22 4979 if (board_config == AD1882_AUTO) e87b24f7 Takashi Iwai 2013-01-22 4980 return ad1882_parse_auto_config(codec); 78bb3cb0 Takashi Iwai 2012-12-21 4981 c5a4bcd0 Takashi Iwai 2009-02-06 4982 err = snd_hda_attach_beep_device(codec, 0x10); c5a4bcd0 Takashi Iwai 2009-02-06 4983 if (err < 0) { c5a4bcd0 Takashi Iwai 2009-02-06 4984 ad198x_free(codec); c5a4bcd0 Takashi Iwai 2009-02-06 4985 return err; c5a4bcd0 Takashi Iwai 2009-02-06 4986 } c5a4bcd0 Takashi Iwai 2009-02-06 @4987 set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); c5a4bcd0 Takashi Iwai 2009-02-06 4988 0ac8551e Takashi Iwai 2007-06-20 4989 spec->multiout.max_channels = 6; 0ac8551e Takashi Iwai 2007-06-20 4990 spec->multiout.num_dacs = 3; 0ac8551e Takashi Iwai 2007-06-20 4991 spec->multiout.dac_nids = ad1882_dac_nids; 0ac8551e Takashi Iwai 2007-06-20 4992 spec->multiout.dig_out_nid = AD1882_SPDIF_OUT; 0ac8551e Takashi Iwai 2007-06-20 4993 spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); 0ac8551e Takashi Iwai 2007-06-20 4994 spec->adc_nids = ad1882_adc_nids; 0ac8551e Takashi Iwai 2007-06-20 4995 spec->capsrc_nids = ad1882_capsrc_nids; --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 1

[alsa-devel] [PATCH 00/12] More updates on HD-audio generic parser
by Takashi Iwai 22 Jan '13

22 Jan '13

Hi, not surprisingly, here is another batch for fixing / improving the HD-audio generic parser code. With these, now AD and IDT codecs have proper analog-loopback features, and VIA and Conexant codecs have better aamix handling. The conflict of aamixer and indep-hp mode seen on VIA codecs should have been fixed now. As usual, all changes are found in test/hda-gen-parser (or test/hda-migrate) branch of sound-unstable tree. It's already merged in master branch as well. Takashi

2 14

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Takashi Iwai 22 Jan '13

22 Jan '13

At Sat, 19 Jan 2013 00:33:52 -0500, Miro Hodak wrote: > > On Fri, Jan 18, 2013 at 1:45 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > OK, thanks for quick tests! > > > > > > Takashi > > > > Testing latest updates in sound-unstable: > > commit f741c8a1f649379f8f3eb81119df2676c46075cf > Merge: d821c1e 164a7ad > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Fri Jan 18 18:30:00 2013 +0100 > > Playback and capture appear to work correctly with one possible caveats: > > Item labeled "Front" controls the level of sound through the speakers. I > would expect that it would only control sound from front HP. This happens > whether Independent HP is on and off. When you look at other controls, there are "Surround", "Center", and "LFE", and these are mixer controls for other two line-out jacks. In that sense, "Front" is corresponding to the same group, the line out jack. You have an individual "Headphone" output. Isn't it obvious? Takashi

3 15

[alsa-devel] [asoc:topic/ioremap 1/1] sound/soc/codecs/jz4740.c:364:2: error: implicit declaration of function 'devm_ioremap_resource'
by kbuild test robot 22 Jan '13

22 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/ioremap head: bc96ea9b07adce5ec678899ea2604bd38677938e commit: bc96ea9b07adce5ec678899ea2604bd38677938e [1/1] ASoC: Convert to devm_ioremap_resource() config: make ARCH=i386 allyesconfig All error/warnings: sound/soc/codecs/jz4740.c: In function 'jz4740_codec_probe': >> sound/soc/codecs/jz4740.c:364:2: error: implicit declaration of function 'devm_ioremap_resource' [-Werror=implicit-function-declaration] sound/soc/codecs/jz4740.c:364:7: warning: assignment makes pointer from integer without a cast [enabled by default] cc1: some warnings being treated as errors vim +/devm_ioremap_resource +364 sound/soc/codecs/jz4740.c 358 jz4740_codec = devm_kzalloc(&pdev->dev, sizeof(*jz4740_codec), 359 GFP_KERNEL); 360 if (!jz4740_codec) 361 return -ENOMEM; 362 363 mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); > 364 base = devm_ioremap_resource(&pdev->dev, mem); 365 if (IS_ERR(base)) 366 return PTR_ERR(base); 367 --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

1 0

[alsa-devel] HDA independent headphone
by Raymond Yau 22 Jan '13

22 Jan '13

> > > Actually, it works fine in Skype when using "AD1989B Analog (hw: 0,2)". I > > > was confused because I changed only "Speakers" to that device and tried > > > making a test sound. That did not work, apparently test sound goes to > > > whatever "Ringing" is set to. But when making actual calls, it works as > > > expected and I can take advantage of the Independent HP feature. > > > > Hm, there should be nothing to touch the indep-HP mixer control from > > PCM, so I have no idea why it went wrong. > > > > BTW, I guess we have still an issue regarding indep HP. When it's ON, > > should the auto-mute be disabled, no? > > Although one can disable the automute manually, it'd be maybe more > > user-friendly if the driver does it by itself. > > > > The patch is below, already merged in test/hda-gen-parser branch (and > test/hda-migrate and master branch, too), as usual. > Does it mean that the value of the "front headphone jack" kcontrol is no longer updated when the independent headphone is switch on ? > > --- > From: Takashi Iwai <tiwai(a)suse.de> > Subject: [PATCH] ALSA: hda - Disable HP auto-mute during independent HP mode > > Both the HP auto-mute and the independent HP mode conflict with each > other. Make HP auto-mute disabled (only for the affected HP jack) > during the driver is in HP independent mode. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > sound/pci/hda/hda_generic.c | 19 ++++++++++++++++--- > 1 file changed, 16 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c > index 63d12ef..258fb5e 100644 > --- a/sound/pci/hda/hda_generic.c > +++ b/sound/pci/hda/hda_generic.c > @@ -1889,6 +1889,13 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol, > *dacp = 0; > else > *dacp = spec->alt_dac_nid; > + > + /* update HP auto-mute state too */ > + if (spec->hp_automute_hook) > + spec->hp_automute_hook(codec, NULL); > + else > + snd_hda_gen_hp_automute(codec, NULL); > + > ret = 1; > } > unlock: > @@ -3467,10 +3474,16 @@ static void call_update_outputs(struct hda_codec *codec) > void snd_hda_gen_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack) > { > struct hda_gen_spec *spec = codec->spec; > + hda_nid_t *pins = spec->autocfg.hp_pins; > + int num_pins = ARRAY_SIZE(spec->autocfg.hp_pins); > + > + /* No detection for the first HP jack during indep-HP mode */ > + if (spec->indep_hp_enabled) { > + pins++; > + num_pins--; > + } > > - spec->hp_jack_present = > - detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), > - spec->autocfg.hp_pins); > + spec->hp_jack_present = detect_jacks(codec, num_pins, pins); > if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo)) > return; > call_update_outputs(codec); > -- > 1.8.1.1 >

2 1

[alsa-devel] [PATCH 33/33] ASoC: Convert to devm_ioremap_resource()
by Thierry Reding 22 Jan '13

22 Jan '13

Convert all uses of devm_request_and_ioremap() to the newly introduced devm_ioremap_resource() which provides more consistent error handling. devm_ioremap_resource() provides its own error messages so all explicit error messages can be removed from the failure code paths. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/cirrus/ep93xx-ac97.c | 7 ++++--- sound/soc/cirrus/ep93xx-i2s.c | 6 +++--- sound/soc/codecs/jz4740.c | 6 +++--- sound/soc/fsl/imx-audmux.c | 6 +++--- sound/soc/fsl/imx-ssi.c | 7 +++---- sound/soc/kirkwood/kirkwood-i2s.c | 8 +++----- sound/soc/mxs/mxs-saif.c | 8 +++----- sound/soc/pxa/mmp-sspa.c | 6 +++--- 8 files changed, 25 insertions(+), 29 deletions(-) diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index f3f50e6..1738d28 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -11,6 +11,7 @@ */ #include <linux/delay.h> +#include <linux/err.h> #include <linux/io.h> #include <linux/init.h> #include <linux/module.h> @@ -367,9 +368,9 @@ static int ep93xx_ac97_probe(struct platform_device *pdev) if (!res) return -ENODEV; - info->regs = devm_request_and_ioremap(&pdev->dev, res); - if (!info->regs) - return -ENXIO; + info->regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(info->regs)) + return PTR_ERR(info->regs); irq = platform_get_irq(pdev, 0); if (!irq) diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 3365d4e..323ed69 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -380,9 +380,9 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) if (!res) return -ENODEV; - info->regs = devm_request_and_ioremap(&pdev->dev, res); - if (!info->regs) - return -ENXIO; + info->regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(info->regs)) + return PTR_ERR(info->regs); info->mclk = clk_get(&pdev->dev, "mclk"); if (IS_ERR(info->mclk)) { diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index d991529..5f607b3 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -361,9 +361,9 @@ static int jz4740_codec_probe(struct platform_device *pdev) return -ENOMEM; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_request_and_ioremap(&pdev->dev, mem); - if (!base) - return -EBUSY; + base = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(base)) + return PTR_ERR(base); jz4740_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base, &jz4740_codec_regmap_config); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 251f4d9..c21ac9c 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -252,9 +252,9 @@ static int imx_audmux_probe(struct platform_device *pdev) of_match_device(imx_audmux_dt_ids, &pdev->dev); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - audmux_base = devm_request_and_ioremap(&pdev->dev, res); - if (!audmux_base) - return -EADDRNOTAVAIL; + audmux_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(audmux_base)) + return PTR_ERR(audmux_base); pinctrl = devm_pinctrl_get_select_default(&pdev->dev); if (IS_ERR(pinctrl)) { diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 3b48042..55464a5 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -550,10 +550,9 @@ static int imx_ssi_probe(struct platform_device *pdev) goto failed_get_resource; } - ssi->base = devm_request_and_ioremap(&pdev->dev, res); - if (!ssi->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; + ssi->base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(ssi->base)) { + ret = PTR_ERR(ssi->base); goto failed_register; } diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 282d8b1..c74c890 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -472,11 +472,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - priv->io = devm_request_and_ioremap(&pdev->dev, mem); - if (!priv->io) { - dev_err(&pdev->dev, "devm_request_and_ioremap failed\n"); - return -ENOMEM; - } + priv->io = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(priv->io)) + return PTR_ERR(priv->io); priv->irq = platform_get_irq(pdev, 0); if (priv->irq <= 0) { diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 365d9d2..b327709 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -724,11 +724,9 @@ static int mxs_saif_probe(struct platform_device *pdev) iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - saif->base = devm_request_and_ioremap(&pdev->dev, iores); - if (!saif->base) { - dev_err(&pdev->dev, "ioremap failed\n"); - return -ENODEV; - } + saif->base = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(saif->base)) + return PTR_ERR(saif->base); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) { diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 41c3a09..9140c4a 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -429,9 +429,9 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) if (res == NULL) return -ENOMEM; - priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res); - if (priv->sspa->mmio_base == NULL) - return -ENODEV; + priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(priv->sspa->mmio_base)) + return PTR_ERR(priv->sspa->mmio_base); priv->sspa->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(priv->sspa->clk)) -- 1.8.1.1

2 3

[alsa-devel] [PATCH] ASoC: wm_adsp: Release firmware on error
by Charles Keepax 22 Jan '13

22 Jan '13

This patch correctly releases the firmware if the magic string in the firmware header does not match. Change-Id: I2d08f7fff92b2f0e0f2ab1fb937007e3783f815e Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4196f2d..b6b6548 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -396,7 +396,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) hdr = (void*)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); - return -EINVAL; + goto out_fw; } adsp_dbg(dsp, "%s: v%d.%d.%d\n", file, -- 1.7.2.5

2 1

[alsa-devel] [PATCH 0/5] ASoC: tegra: Some fixes in codec files
by Sachin Kamat 22 Jan '13

22 Jan '13

Compile tested using tegra_defconfig on the for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Sachin Kamat (5): ASoC: tegra_wm9712: Remove __devinitconst attribute ASoC: tegra20_ac97: Remove __devinitconst attribute ASoC: tegra: Make 'ahub_auxdata' static ASoC: tegra: Use NULL instead of 0 for pointers ASoC: tegra: Staticize some functions in tegra30_i2s.c sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 6 +++--- sound/soc/tegra/tegra30_i2s.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 2 +- 4 files changed, 7 insertions(+), 7 deletions(-) -- 1.7.4.1

2 9

[alsa-devel] [PATCH 1/2] ALSA: hda - Add fixup for Acer AO725 laptop
by Takashi Iwai 21 Jan '13

21 Jan '13

Acer AO725 needs the same fixup as AO756. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- it's targeted for 3.8-rc5 inclusion sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6369b23..cf38861 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6266,6 +6266,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), -- 1.8.1.1

1 0

[alsa-devel] [PATCH 2/2] ALSA: hda - Add Conexant CX20755/20756/20757 codec IDs
by Takashi Iwai 21 Jan '13

21 Jan '13

These are just compatible with other CX2075x codecs. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- it's targeted for 3.8-rc5 inclusion sound/pci/hda/patch_conexant.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index dd798c3..009b77a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4636,6 +4636,12 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f15111, .name = "CX20753/4", .patch = patch_conexant_auto }, + { .id = 0x14f15113, .name = "CX20755", + .patch = patch_conexant_auto }, + { .id = 0x14f15114, .name = "CX20756", + .patch = patch_conexant_auto }, + { .id = 0x14f15115, .name = "CX20757", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -4659,6 +4665,9 @@ MODULE_ALIAS("snd-hda-codec-id:14f150b9"); MODULE_ALIAS("snd-hda-codec-id:14f1510f"); MODULE_ALIAS("snd-hda-codec-id:14f15110"); MODULE_ALIAS("snd-hda-codec-id:14f15111"); +MODULE_ALIAS("snd-hda-codec-id:14f15113"); +MODULE_ALIAS("snd-hda-codec-id:14f15114"); +MODULE_ALIAS("snd-hda-codec-id:14f15115"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.8.1.1

1 0

[alsa-devel] Invitation to connect on LinkedIn
by Putinei Ionut via LinkedIn 21 Jan '13

21 Jan '13

LinkedIn ------------ Putinei Ionut requested to add you as a connection on LinkedIn: ------------------------------------------ abdul, Find out why I use LinkedIn. Stay in touch and build your professional network. - Putinei Accept invitation from Putinei Ionut http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/uii-rtptP1H4ViCg9YiBUOvto1r4_5… View profile of Putinei Ionut http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/rso/227244299/bCFj/name/138036… ------------------------------------------ You are receiving Invitation emails. This email was intended for abdul moiz. Learn why this is included: http://www.linkedin.com/e/jpxkk3-hc7ogynz-70/plh/http%3A%2F%2Fhelp%2Elinked… (c) 2012, LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

[alsa-devel] [PATCH] ASoC: arizona: Support clearing clocks
by Mark Brown 21 Jan '13

21 Jan '13

Some systems may wish to support switching between telephony and CD audio clock rates but this is restricted by enforcement of constraints on the current DAI clock. Support setting clocks to zero and don't enforce any constraints in that case in order to facilitate this use case. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 4c2b626..8c56488 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -585,6 +585,10 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, case 147456000: val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT; break; + case 0: + dev_dbg(arizona->dev, "%s cleared\n", name); + *clk = freq; + return 0; default: return -EINVAL; } @@ -803,6 +807,9 @@ static int arizona_startup(struct snd_pcm_substream *substream, return 0; } + if (base_rate == 0) + return 0; + if (base_rate % 8000) constraint = &arizona_44k1_constraint; else -- 1.7.10.4

1 0

[alsa-devel] question on snd_soc_dai_link_event
by Sebastien LEDUC 21 Jan '13

21 Jan '13

Hi, I have one question about current implementation of function snd_soc_dai_link_event Currently, on power up, it calls hw_params on the source and sink dai drivers, and digital_mute(0) on the sink dai driver I wonder why it does not call the full hw_params / prepare / trigger sequence on both dai drivers to start the 2 DAIs. Thanks in advance Regards, Sebastien

2 5

[alsa-devel] Focusrite Scarlett 18i6 mixer
by Robin Gareus 20 Jan '13

20 Jan '13

Hi, I've started to reverse engineer the 18i6 mixer protocol and came up with a proof-of-concept test script which includes documentation: https://github.com/x42/scarlettmixer Except for a few fixed controls and switches, the 18i6's mixer and router is [likely] too complex to be exposed via ALSA's mixer interface. AFAICT it'll require a dedicated control application - like hdspmixer - in order to make full use of it. The current proof-of-concept uses pyUSB to send URB control messages and therefore requires root access. It will also not work while the device is in use (by snd-usb-audio). A dedicated ioctl interface - similar to SNDRV_HDSP_ - should be added to ALSA. However, it is yet unclear to me how much of the protocol should be moved to kernel-space. Anyway, the script can be used to configure the device, save settings to it and subsequently use those settings with snd-usb-audio and Martin's patch. Thoughts, comments? robin PS. the same protocol may apply to other Focusrite Scalett USB2 devices such as the 8i6 - some of the factory-reset traces hint at that, but I would not know.

1 1

[alsa-devel] Must MIDI SysEx data contain the start byte 0xf0 and the end byte 0xf7?
by Joerg-Cyril.Hoehle＠t-systems.com 20 Jan '13

20 Jan '13

Hi, I've read about SysEx continuations aka. divided SysEx http://www.allegro.cc/forums/thread/607445 http://www.sonicspot.com/guide/midifiles.html What it means is that within a MIDI file, a long sysex could be split across several events with different delta times, giving the receiver some processing time in between (e.g. flash EEPROM etc.). The delta times tell when to send the parts. Now, is that truth? How to do that with ALSA? The documentation on snd_seq_ev_set_sysex insists: "the sysex data must contain the start byte 0xf0 and the end byte 0xf7." http://www.alsa-project.org/alsa-doc/alsa-lib/group___seq_middle.html#ga104… Would it be the sole business of the low-level ALSA driver to slowly send large (and complete) sysex data, out of application control (e.g. the delta times would get lost)? Should apps always coalesce partial sysex data and hand out the complete blob to ALSA's snd_seq_ev_set_sysex? Is divided SysEx too rare to worry about? I'm asking because we've seen a user of Wine & KORG stumble upon a non-F7-terminated SysEx. Regards, Jörg Höhle

2 3

[alsa-devel] [PATCH] ASoC: core: Ensure SND_SOC_BYTES writes are from DMA safe memory
by Mark Brown 20 Jan '13

20 Jan '13

With some buses the transfers may DMAed, especially for larger blocks. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 625e973..3e4e40e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3131,9 +3131,12 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (!codec->using_regmap) return -EINVAL; - data = ucontrol->value.bytes.data; len = params->num_regs * codec->val_bytes; + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + /* * If we've got a mask then we need to preserve the register * bits. We shouldn't modify the incoming data so take a @@ -3146,10 +3149,6 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, val &= params->mask; - data = kmemdup(data, len, GFP_KERNEL); - if (!data) - return -ENOMEM; - switch (codec->val_bytes) { case 1: ((u8 *)data)[0] &= ~params->mask; @@ -3171,8 +3170,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(codec->control_data, params->base, data, len); - if (params->mask) - kfree(data); + kfree(data); return ret; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm2200: Implement EQ and LHPF coefficient configuration
by Mark Brown 20 Jan '13

20 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2200.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 732b3dd..1be61e6 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1141,6 +1141,12 @@ SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_3L, WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SND_SOC_BYTES_MASK("EQL Coefficients", WM2200_EQL_1, 20, WM2200_EQL_ENA), +SND_SOC_BYTES_MASK("EQR Coefficients", WM2200_EQR_1, 20, WM2200_EQR_ENA), + +SND_SOC_BYTES("LHPF1 Coefficeints", WM2200_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficeints", WM2200_HPLPF2_2, 1), + SOC_SINGLE("OUT1 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_1L, WM2200_OUT1_OSR_SHIFT, 1, 0), SOC_SINGLE("OUT2 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_2L, -- 1.7.10.4

1 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Make region identification errors more informative
by Mark Brown 20 Jan '13

20 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f956c35..267a6f1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -773,7 +773,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) break; default: - adsp_err(dsp, "Unknown region type %x\n", type); + adsp_err(dsp, "%s.%d: Unknown region type %x at %d\n", + file, blocks, type, pos); break; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Use GFP_DMA for things that may be DMAed
by Mark Brown 20 Jan '13

20 Jan '13

Normally kmalloc() returns things that are DMA safe so not visible on all platforms but we do need to explicitly request DMA safe memory. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f263bfd..f956c35 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -410,7 +410,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) if (reg) { buf = kmemdup(region->data, le32_to_cpu(region->len), - GFP_KERNEL); + GFP_KERNEL | GFP_DMA); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; @@ -557,7 +557,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) adsp_warn(dsp, "Algorithm list end %x 0x%x != 0xbeadead\n", term, be32_to_cpu(val)); - alg = kzalloc((term - pos) * 2, GFP_KERNEL); + alg = kzalloc((term - pos) * 2, GFP_KERNEL | GFP_DMA); if (!alg) return -ENOMEM; @@ -779,7 +779,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) if (reg) { buf = kmemdup(blk->data, le32_to_cpu(blk->len), - GFP_KERNEL); + GFP_KERNEL | GFP_DMA); if (!buf) { adsp_err(dsp, "Out of memory\n"); return -ENOMEM; -- 1.7.10.4

1 0

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Takashi Iwai 19 Jan '13

19 Jan '13

At Thu, 17 Jan 2013 21:40:43 +0100, Manolo Díaz wrote: > > El jue, 17 ene 2013 a las 21:16 horas > Takashi Iwai escribió: > > >At Thu, 17 Jan 2013 19:12:24 +0100, > >Manolo Díaz wrote: > >> > >> El jue, 17 ene 2013 a las 16:32 horas > >> Takashi Iwai escribió: > >> > >> >At Thu, 17 Jan 2013 09:59:20 -0500, > >> >Miro Hodak wrote: > >> >> > >> >> On Thu, Jan 17, 2013 at 7:49 AM, Raymond Yau <superquad.vortex2(a)gmail.com>wrote: > >> >> > >> >> > > >> >> > > > >> >> > > OK, now as far as Auto-Mute goes, should it do what expected when > >> >> > enabled - i.e. muting the speakers when headphones are connected to the > >> >> > front panel? I have tried that, does not seem to work. > >> >> > > > >> >> > > >> >> > Do your computer chassis have a hda front audio panel with Jack detection ? > >> >> > > >> >> > The front headphone Jack should return true when headphone Jack is plugged > >> >> > > >> >> > control.35 { > >> >> > iface CARD > >> >> > name 'Front Headphone Jack' > >> >> > > >> >> > value false > >> >> > comment { > >> >> > access read > >> >> > type BOOLEAN > >> >> > count 1 > >> >> > } > >> >> > } > >> >> > > >> >> > All ADI codecs support impedance measurement > >> >> > > >> >> > Try the modified hda-jack-sense-test which need hda-analyzer > >> >> > > >> >> > http://thread.gmane.org/gmane.linux.alsa.devel/90911/focus=91115 > >> >> > > >> >> > >> >> Hmmm, ran into some python issues. Even hda-analyzer itself is not working: > >> >> ... > >> >> File "/dev/shm/hda-analyzer/hda_proc.py", line 143, in wrongfile > >> >> raise ValueError, "wrong proc file format (%s)" % msg > >> >> ValueError: wrong proc file format (integer decode 'State of AFG node > >> >> 0x01:' ('Node ')) > >> > > >> >Ah, it's a new proc file entry that has been added recently. > >> >hda-analyzer needs to be fixed. > >> > > >> >In anyway, only for checking the presence unsolicited events, you can > >> >do without extra program. Suppose you built your kernel with the > >> >tracing support. See "Tracepoints" option in > >> >Documentation/sound/alsa/HD-Audio.txt. > >> > > >> > # echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable > >> > > >> > (plug and unplug your headphone jacks) > >> > > >> > # cat /sys/kernel/debug/tracing/trace > >> > > >> >If you see some hda_unsol_event lines there, it implies that the jack > >> >detection basically works in hardware level. > >> > > >> > > >> >Takashi > >> > >> I Suppose that when auto-mute is enabled an HP independent is disabled, > >> inserting the HP jack should disable sound from front speakers. If so > >> then auto-mute doesn't work. Jack insertion/extraction events are well > >> detected for front speakers and HP. > >> > >> #~ cat /sys/kernel/debug/tracing/trace > >> # tracer: nop > >> # > >> # entries-in-buffer/entries-written: 4/4 #P:2 > >> # > >> # _-----=> irqs-off > >> # / _----=> need-resched > >> # | / _---=> hardirq/softirq > >> # || / _--=> preempt-depth > >> # ||| / delay > >> # TASK-PID CPU# |||| TIMESTAMP FUNCTION > >> # | | | |||| | | > >> <idle>-0 [001] d.h. 137.881919: hda_unsol_event: [0] res=8000000, res_ex=10 > >> <idle>-0 [001] d.h. 162.511102: hda_unsol_event: [0] res=8000000, res_ex=10 > >> <idle>-0 [001] d.h. 246.529813: hda_unsol_event: [0] res=4000000, res_ex=10 > >> <idle>-0 [001] d.h. 272.381719: hda_unsol_event: [0] res=4000000, res_ex=10 > >> > >> branch tested: test/hda-gen-parser > > > >Please test with either master or test/hda-migrate branch. > >test/hda-gen-parser branch doesn't contain the patch for AD codecs. > > > >In anyway, please attach alsa-info.sh outputs before and after > >plugging the headphone with the latest sound-unstable tree (and a > >proper branch). > > > > > >thanks, > > > >Takashi > > branch tested: master > > commit d89012bef4ad6d33c1050dd2924f010f5072ff52 > Merge: ad92fbe 36c9db7 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Jan 17 17:47:19 2013 +0100 > > Just the same: auto-mute doesn't work. > > alsa-info output show no differences (attached). Thanks. It turned out that simply missing unsol_event handlers in the codec driver code. The patch below should fix the issue. It's already merged in test/hda-migrate and master branches. Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix missing unsol event handler in some codec drivers This resulted in non-working auto-mute behavior, of course... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_analog.c | 1 + sound/pci/hda/patch_ca0110.c | 1 + sound/pci/hda/patch_cmedia.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 02fe0d1..6feaec4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -595,6 +595,7 @@ static const struct hda_codec_ops ad198x_auto_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = ad198x_free, + .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, .suspend = ad198x_suspend, diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 8d09325..db7635c 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -36,6 +36,7 @@ static const struct hda_codec_ops ca0110_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = snd_hda_gen_free, + .unsol_event = snd_hda_jack_unsol_event, }; static int ca0110_parse_auto_config(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 04dd3b6..087cabb 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -576,6 +576,7 @@ static const struct hda_codec_ops cmi_auto_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, .free = snd_hda_gen_free, + .unsol_event = snd_hda_jack_unsol_event, }; static int cmi_parse_auto_config(struct hda_codec *codec) -- 1.8.1

4 17

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Raymond Yau 19 Jan '13

19 Jan '13

> > > Testing latest updates in sound-unstable: > > commit f741c8a1f649379f8f3eb81119df2676c46075cf > Merge: d821c1e 164a7ad > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Fri Jan 18 18:30:00 2013 +0100 > > Playback and capture appear to work correctly with one possible caveats: > > Item labeled "Front" controls the level of sound through the speakers. I would expect that it would only control sound from front HP. This happens whether Independent HP is on and off. > Is it feasible to add a hint which disable the creation of virtual master playback volume and switch ?

2 1

[alsa-devel] [PATCH 0/3] You guessed it, it's generic parser patches
by David Henningsson 18 Jan '13

18 Jan '13

The third one seems to apply to the regular sound tree too. David Henningsson (3): ALSA: hda - make sure there are enough input labels and paths ALSA: hda - don't compare with yourself in fill_input_pin_labels ALSA: hda - Don't add unnecessary indices on HDMI and SPDIF sound/pci/hda/hda_auto_parser.c | 22 ++++++++++++---------- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_generic.h | 6 +++--- 3 files changed, 17 insertions(+), 15 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ALSA: hda - Fix mute led for another HP machine
by David Henningsson 18 Jan '13

18 Jan '13

This machine also has the "HP_Mute_LED_0_A" string in DMI information. Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1096789 Tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f519627..6369b23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6251,6 +6251,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1977, "HP Pavilion 14", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/4] Fixes for missing boost controls in generic parser
by Takashi Iwai 18 Jan '13

18 Jan '13

Hi, this is a series of patches to fix the missing boost controls in the generic parser. I found the problem not only on AD1988 but some others, and while debugging this issue, more issues have been revealed, thus it became more volumes than expected. All patches are found in sound-unstable tree test/hda-gen-parser branch. And, now both test/hda-gen-parser branch and test/hda-migrate branch become identical. It doesn't matter which branch you take. Also, it's already merged in master branch of sound-unstable tree. So.. at this point, it'd be helpful if anyone can test this tree, and let me know any issues. thanks, Takashi

2 6

Re: [alsa-devel] [PATCH] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Hebbar, Gururaja 18 Jan '13

18 Jan '13

On Fri, Jan 18, 2013 at 14:13:40, Michal Bachraty wrote: > On Friday, January 18, 2013 08:22:53 Hebbar, Gururaja wrote: > > On Tue, Jan 15, 2013 at 20:11:00, Michal Bachraty wrote: > > > Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> > > > > Even simple patches need a simple changelog. > > Add a brief description about what & why it is done? > > > > Hi Gurujaja, > > > > - u32 rotate = (32 - word_length) / 4; > This implementation is wrong, but it works only for 16, or 32 bit audio data. > (rotation for 16 or 32 bit is same as in code I present) Mcasp rotated data in > 4 bits (max value 0x7)and then masks them . That data are sended to i2s bus. > For 24 bit or 20 bit or other data formats, this code rotates data badly and > you hear somethink like noise. You need to use > > > + u32 rotate = (word_length / 4) & 0x7; > to proper data rotation. Correct. Now can you add the above info into the patch and re-submit. That way when users sees this patch, they will get all the info needed. > > Regards, > Michal > > Regards, Gururaja

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock
by Chris Rattray 18 Jan '13

18 Jan '13

Sets the ADSP1 clock rate to match the system clock rate. To support this the codec driver provides details of register containing the system clock control bits. Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 +++ 2 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9a3f220..56bd6ba 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -103,6 +103,13 @@ #define ADSP1_START_SHIFT 0 /* DSP1_START */ #define ADSP1_START_WIDTH 1 /* DSP1_START */ +/* + * ADSP1 Control 31 + */ +#define ADSP1_CLK_SEL_MASK 0x0007 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_SHIFT 0 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_WIDTH 3 /* CLK_SEL_ENA */ + #define ADSP2_CONTROL 0 #define ADSP2_CLOCKING 1 #define ADSP2_STATUS1 4 @@ -817,12 +824,38 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; int ret; + int val; switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, ADSP1_SYS_ENA); + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + if(dsp->sysclk_reg) { + ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", + ret); + return ret; + } + + val = (val & dsp->sysclk_mask) + >> dsp->sysclk_shift; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP1_CONTROL_31, + ADSP1_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", + ret); + return ret; + } + } + ret = wm_adsp_load(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41206d7..cb8871a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -40,6 +40,9 @@ struct wm_adsp { struct regmap *regmap; int base; + int sysclk_reg; + int sysclk_mask; + int sysclk_shift; struct list_head alg_regions; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback
by Michal Bachraty 18 Jan '13

18 Jan '13

Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 13f8d73..a03bece 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -626,7 +626,7 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (32 - word_length) / 4; + u32 rotate = (word_length / 4) & 0x7; u32 mask = (1ULL << word_length) - 1; /* -- 1.7.9.5

2 1

[alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Miro Hodak 18 Jan '13

18 Jan '13

Commit identified in the subject (ALSA: HDA - Add Independent Headphone for all models of ad1988/ad1989) causes problem with audio on at least two Asus boards and is still present in current kernel. The boards on which this problem is seen are Asus M3A79-T DELUXE and Asus Rampage II Gene (actually a slightly modified version used in Asus CG5290 desktop). The symptoms: 1. XFCE and Gnome2 audio mixers crash whenever sound is started or stopped. The error message is: simple_none.c:1350: get_enum_item_ops: Assertion `helem' failed. 2. On one of the boards (Asus M3A79-T DELUXE) the mixer control labels are misconfigured since muting the Master channel does not turn the sound off. Either Master is mislabeled or it does not work correctly. Other than that, the sound actually works fine, but it is very annoying when mixer keeps crashing. Reverting to kernel just before the commit fixes these issues. Thus it appears that the commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4 breaks autodetect for these boards. A workaround is to force the 6stack-dig model and this fixes problems on both boards, but, of course, it would be preferred if things work out of the box. We do not understand enough of alsa in kernel to propose a patch, but hope that we have provided enough information that it will be easy to write. We are happy to provide more information and test any proposed patches. Alsa-info for both boards can be found below. Best regards, Miro Hodak and Manolo Diaz Results of alsa-info Asus Rampage II Gene: Before the commit: http://pastebin.com/gCZurk6D After the commit (autodetect): http://pastebin.com/3hNX2g1c After the commit (6stack_dig model): http://pastebin.com/dL0UGeTW More recent kernel: 3.6.11-gentoo (autodetect): http://pastebin.com/0gjnhYA2 3.6.11-gentoo (6stack_dig model): http://pastebin.com/K9bsWBWW Asus M3A79-T DELUXE Before the commit: http://pastebin.com/hyUBmpDL After the commit (autodetect) : http://pastebin.com/gzGg60bM After the commit (6stack_dig model): http://pastebin.com/y88jCMhM

7 53

[alsa-devel] Problems capturing and playback via the loopback device
by Erik Kazandjian 18 Jan '13

18 Jan '13

Hi, I'm new to alsa and I'm trying to do the following. I have configured my rythm box to use the loopback device. I've written a little program to capture everything that comes on the other side of the loopback and save that into a file. If I play a song on rythm box I see actual packets on the couterside of the loopback however I save these packets in a file and try to play them back using my sound card I hear no sound. I use the following .asoundrc pcm.loopplayback { type hw card 1 device 1 subdevice 0 } pcm.loopcapture { type hw card 1 device 0 subdevice 0 } pcm.play48 { type plug slave { pcm loopplayback rate 44100 channels 2 } } attached is my test program to do the capturing. Anyone has any idea what I'm doing wrong ? Kind regards Erik

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock
by Chris Rattray 18 Jan '13

18 Jan '13

Sets the ADSP1 clock rate to match the system clock rate. To support this the codec driver provides details of register containing the system clock control bits. Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 35 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 +++ 2 files changed, 38 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9a3f220..739dab6 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -75,6 +75,8 @@ #define ADSP1_CONTROL_30 0x24 #define ADSP1_CONTROL_31 0x26 +#define ADSP1_CLOCKING 38 + /* * ADSP1 Control 19 */ @@ -103,6 +105,13 @@ #define ADSP1_START_SHIFT 0 /* DSP1_START */ #define ADSP1_START_WIDTH 1 /* DSP1_START */ +/* + * ADSP1 Control 31 + */ +#define ADSP1_CLK_SEL_MASK 0x0007 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_SHIFT 0 /* CLK_SEL_ENA */ +#define ADSP1_CLK_SEL_WIDTH 3 /* CLK_SEL_ENA */ + #define ADSP2_CONTROL 0 #define ADSP2_CLOCKING 1 #define ADSP2_STATUS1 4 @@ -817,12 +826,38 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; int ret; + int val; switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, ADSP1_SYS_ENA); + /* + * For simplicity set the DSP clock rate to be the + * SYSCLK rate rather than making it configurable. + */ + if(dsp->sysclk_reg) { + ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); + if (ret != 0) { + adsp_err(dsp, "Failed to read SYSCLK state: %d\n", + ret); + return ret; + } + + val = (val & dsp->sysclk_mask) + >> dsp->sysclk_shift; + + ret = regmap_update_bits(dsp->regmap, + dsp->base + ADSP1_CLOCKING, + ADSP1_CLK_SEL_MASK, val); + if (ret != 0) { + adsp_err(dsp, "Failed to set clock rate: %d\n", + ret); + return ret; + } + } + ret = wm_adsp_load(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 41206d7..cb8871a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -40,6 +40,9 @@ struct wm_adsp { struct regmap *regmap; int base; + int sysclk_reg; + int sysclk_mask; + int sysclk_shift; struct list_head alg_regions; -- 1.7.2.5

2 2

Re: [alsa-devel] HD-audio regression after commit 34588709af61be1550b4e2bcee5c85d0ac4f34d4
by Miro Hodak 18 Jan '13

18 Jan '13

On Thu, Jan 17, 2013 at 3:40 PM, Manolo Díaz <diaz.manolo(a)gmail.com> wrote: > > branch tested: master > > commit d89012bef4ad6d33c1050dd2924f010f5072ff52 > Merge: ad92fbe 36c9db7 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Thu Jan 17 17:47:19 2013 +0100 > > Just the same: auto-mute doesn't work. > > alsa-info output show no differences (attached). > > Jack inserting event correctly detected again. > > #~ cat /sys/kernel/debug/tracing/trace > # tracer: nop > # > # entries-in-buffer/entries-written: 1/1 #P:2 > # > # _-----=> irqs-off > # / _----=> need-resched > # | / _---=> hardirq/softirq > # || / _--=> preempt-depth > # ||| / delay > # TASK-PID CPU# |||| TIMESTAMP FUNCTION > # | | | |||| | | > <idle>-0 [001] d.h. 198.843063: hda_unsol_event: [0] > res=4000000, res_ex=10 > > > Regards, > -- > Manolo Díaz > Testing master branch, I got the similar results as Manolo commit 920544e4bc95d5fb4465eb58e419a2c61074cd7e Merge: 910fb91 ae12700 Author: Takashi Iwai <tiwai(a)suse.de> Date: Wed Jan 16 16:26:03 2013 +0100 # cat /sys/kernel/debug/tracing/trace # tracer: nop # # entries-in-buffer/entries-written: 1/1 #P:8 # # _-----=> irqs-off # / _----=> need-resched # | / _---=> hardirq/softirq # || / _--=> preempt-depth # ||| / delay # TASK-PID CPU# |||| TIMESTAMP FUNCTION # | | | |||| | | <idle>-0 [000] d.h. 341.307150: hda_unsol_event: [0] res=4000000, res_ex=10 No difference in alsa-info before and after headphones are plugged in (other than timestamp): http://pastebin.com/J1PiRia8 Should I test with the test/hda-gen-parser branch?

1 0

[alsa-devel] [PATCH RFC] ALSA: hda - Use generic parser for STAC/IDT codec driver
by Takashi Iwai 17 Jan '13

17 Jan '13

Finally we reached here. All codecs driver (except for CA0132, which has really device-specific requirements) have been converted to use the generic parser. This patch appears bigger than others since it also involves with the code shuffling, but mostly the cut-off of parser codes and adapt to the generic parser flags. Most of fixup codecs haven't been changed but just removed a few unnecessary codes. The only missing stuff is the SPDIF mux control. It'll be added again later. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 1 + sound/pci/hda/patch_sigmatel.c | 5219 ++++++++-------------------------------- 2 files changed, 1019 insertions(+), 4201 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 4004d40..4466bf6 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -112,6 +112,7 @@ config SND_HDA_CODEC_ANALOG config SND_HDA_CODEC_SIGMATEL bool "Build IDT/Sigmatel HD-audio codec support" default y + select SND_HDA_GENERIC help Say Y here to include IDT (Sigmatel) HD-audio codec support in snd-hda-intel driver, such as STAC9200. diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f975735..9d2dfad 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -39,14 +39,11 @@ #include "hda_auto_parser.h" #include "hda_beep.h" #include "hda_jack.h" +#include "hda_generic.h" enum { - STAC_VREF_EVENT = 1, - STAC_INSERT_EVENT, + STAC_VREF_EVENT = 8, STAC_PWR_EVENT, - STAC_HP_EVENT, - STAC_LO_EVENT, - STAC_MIC_EVENT, }; enum { @@ -115,7 +112,6 @@ enum { STAC_HP_DV4, STAC_HP_DV5, STAC_HP_HDX, - STAC_HP_DV4_1222NR, STAC_92HD71BXX_HP, STAC_92HD71BXX_NO_DMIC, STAC_92HD71BXX_NO_SMUX, @@ -173,30 +169,14 @@ enum { STAC_9872_MODELS }; -struct sigmatel_mic_route { - hda_nid_t pin; - signed char mux_idx; - signed char dmux_idx; -}; - -#define MAX_PINS_NUM 16 -#define MAX_ADCS_NUM 4 -#define MAX_DMICS_NUM 4 - struct sigmatel_spec { - struct snd_kcontrol_new *mixers[4]; - unsigned int num_mixers; + struct hda_gen_spec gen; unsigned int eapd_switch: 1; - unsigned int surr_switch: 1; - unsigned int alt_switch: 1; - unsigned int hp_detect: 1; - unsigned int spdif_mute: 1; - unsigned int check_volume_offset:1; - unsigned int auto_mic:1; unsigned int linear_tone_beep:1; unsigned int headset_jack:1; /* 4-pin headset jack (hp + mono mic) */ unsigned int volknob_init:1; /* special volume-knob initialization */ + unsigned int powerdown_adcs:1; /* gpio lines */ unsigned int eapd_mask; @@ -218,6 +198,7 @@ struct sigmatel_spec { /* analog loopback */ const struct snd_kcontrol_new *aloopback_ctl; + unsigned int aloopback; unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -225,458 +206,130 @@ struct sigmatel_spec { unsigned int power_map_bits; unsigned int num_pwrs; const hda_nid_t *pwr_nids; - const hda_nid_t *dac_list; - - /* playback */ - struct hda_input_mux *mono_mux; - unsigned int cur_mmux; - struct hda_multi_out multiout; - hda_nid_t dac_nids[5]; - hda_nid_t hp_dacs[5]; - hda_nid_t speaker_dacs[5]; - - int volume_offset; - - /* capture */ - const hda_nid_t *adc_nids; - unsigned int num_adcs; - const hda_nid_t *mux_nids; - unsigned int num_muxes; - const hda_nid_t *dmic_nids; - unsigned int num_dmics; - const hda_nid_t *dmux_nids; - unsigned int num_dmuxes; - const hda_nid_t *smux_nids; - unsigned int num_smuxes; - unsigned int num_analog_muxes; - - const unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */ - const unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */ - unsigned int num_caps; /* number of capture volume/switch elements */ - - struct sigmatel_mic_route ext_mic; - struct sigmatel_mic_route int_mic; - struct sigmatel_mic_route dock_mic; - - const char * const *spdif_labels; - - hda_nid_t dig_in_nid; - hda_nid_t mono_nid; + unsigned int active_adcs; + + /* beep widgets */ hda_nid_t anabeep_nid; hda_nid_t digbeep_nid; - - /* pin widgets */ - const hda_nid_t *pin_nids; - unsigned int num_pins; - - /* codec specific stuff */ - const struct hda_verb *init; - const struct snd_kcontrol_new *mixer; - - /* capture source */ - struct hda_input_mux *dinput_mux; - unsigned int cur_dmux[2]; - struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; - struct hda_input_mux *sinput_mux; - unsigned int cur_smux[2]; - unsigned int cur_amux; - hda_nid_t *amp_nids; - unsigned int powerdown_adcs; - - /* i/o switches */ - unsigned int io_switch[2]; - unsigned int clfe_swap; - hda_nid_t line_switch; /* shared line-in for input and output */ - hda_nid_t mic_switch; /* shared mic-in for input and output */ - hda_nid_t hp_switch; /* NID of HP as line-out */ - unsigned int aloopback; - - struct hda_pcm pcm_rec[2]; /* PCM information */ - - /* dynamic controls and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_dimux; - struct hda_input_mux private_imux; - struct hda_input_mux private_smux; - struct hda_input_mux private_mono_mux; - - /* auto spec */ - unsigned auto_pin_cnt; - hda_nid_t auto_pin_nids[MAX_PINS_NUM]; - unsigned auto_adc_cnt; - hda_nid_t auto_adc_nids[MAX_ADCS_NUM]; - hda_nid_t auto_mux_nids[MAX_ADCS_NUM]; - hda_nid_t auto_dmux_nids[MAX_ADCS_NUM]; - unsigned long auto_capvols[MAX_ADCS_NUM]; - unsigned auto_dmic_cnt; - hda_nid_t auto_dmic_nids[MAX_DMICS_NUM]; - - struct hda_vmaster_mute_hook vmaster_mute; }; #define AC_VERB_IDT_SET_POWER_MAP 0x7ec #define AC_VERB_IDT_GET_POWER_MAP 0xfec -static const hda_nid_t stac9200_adc_nids[1] = { - 0x03, -}; - -static const hda_nid_t stac9200_mux_nids[1] = { - 0x0c, -}; - -static const hda_nid_t stac9200_dac_nids[1] = { - 0x02, -}; - static const hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0a, 0x0b, 0x0c, 0xd, 0x0e, 0x0f, 0x10, 0x11 }; -static const hda_nid_t stac92hd73xx_slave_dig_outs[2] = { - 0x26, 0, -}; - -static const hda_nid_t stac92hd73xx_adc_nids[2] = { - 0x1a, 0x1b -}; - -#define STAC92HD73XX_NUM_DMICS 2 -static const hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { - 0x13, 0x14, 0 -}; - -#define STAC92HD73_DAC_COUNT 5 - -static const hda_nid_t stac92hd73xx_mux_nids[2] = { - 0x20, 0x21, -}; - -static const hda_nid_t stac92hd73xx_dmux_nids[2] = { - 0x20, 0x21, -}; - -static const hda_nid_t stac92hd73xx_smux_nids[2] = { - 0x22, 0x23, -}; - -#define STAC92HD73XX_NUM_CAPS 2 -static const unsigned long stac92hd73xx_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x20, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), -}; -#define stac92hd73xx_capsws stac92hd73xx_capvols - -#define STAC92HD83_DAC_COUNT 3 - static const hda_nid_t stac92hd83xxx_pwr_nids[7] = { 0x0a, 0x0b, 0x0c, 0xd, 0x0e, 0x0f, 0x10 }; -static const hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { - 0x1e, 0, -}; - -static const hda_nid_t stac92hd83xxx_dmic_nids[] = { - 0x11, 0x20, -}; - static const hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f }; -static const hda_nid_t stac92hd71bxx_adc_nids[2] = { - 0x12, 0x13, -}; - -static const hda_nid_t stac92hd71bxx_mux_nids[2] = { - 0x1a, 0x1b -}; - -static const hda_nid_t stac92hd71bxx_dmux_nids[2] = { - 0x1c, 0x1d, -}; - -static const hda_nid_t stac92hd71bxx_smux_nids[2] = { - 0x24, 0x25, -}; - -#define STAC92HD71BXX_NUM_DMICS 2 -static const hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { - 0x18, 0x19, 0 -}; - -static const hda_nid_t stac92hd71bxx_dmic_5port_nids[STAC92HD71BXX_NUM_DMICS] = { - 0x18, 0 -}; - -static const hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { - 0x22, 0 -}; - -#define STAC92HD71BXX_NUM_CAPS 2 -static const unsigned long stac92hd71bxx_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), -}; -#define stac92hd71bxx_capsws stac92hd71bxx_capvols - -static const hda_nid_t stac925x_adc_nids[1] = { - 0x03, -}; - -static const hda_nid_t stac925x_mux_nids[1] = { - 0x0f, -}; - -static const hda_nid_t stac925x_dac_nids[1] = { - 0x02, -}; - -#define STAC925X_NUM_DMICS 1 -static const hda_nid_t stac925x_dmic_nids[STAC925X_NUM_DMICS + 1] = { - 0x15, 0 -}; - -static const hda_nid_t stac925x_dmux_nids[1] = { - 0x14, -}; - -static const unsigned long stac925x_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT), -}; -static const unsigned long stac925x_capsws[] = { - HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), -}; - -static const hda_nid_t stac922x_adc_nids[2] = { - 0x06, 0x07, -}; - -static const hda_nid_t stac922x_mux_nids[2] = { - 0x12, 0x13, -}; - -#define STAC922X_NUM_CAPS 2 -static const unsigned long stac922x_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), -}; -#define stac922x_capsws stac922x_capvols - -static const hda_nid_t stac927x_slave_dig_outs[2] = { - 0x1f, 0, -}; - -static const hda_nid_t stac927x_adc_nids[3] = { - 0x07, 0x08, 0x09 -}; - -static const hda_nid_t stac927x_mux_nids[3] = { - 0x15, 0x16, 0x17 -}; - -static const hda_nid_t stac927x_smux_nids[1] = { - 0x21, -}; - -static const hda_nid_t stac927x_dac_nids[6] = { - 0x02, 0x03, 0x04, 0x05, 0x06, 0 -}; - -static const hda_nid_t stac927x_dmux_nids[1] = { - 0x1b, -}; - -#define STAC927X_NUM_DMICS 2 -static const hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { - 0x13, 0x14, 0 -}; - -#define STAC927X_NUM_CAPS 3 -static const unsigned long stac927x_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x19, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_INPUT), -}; -static const unsigned long stac927x_capsws[] = { - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), -}; - -static const char * const stac927x_spdif_labels[5] = { - "Digital Playback", "ADAT", "Analog Mux 1", - "Analog Mux 2", "Analog Mux 3" -}; - -static const hda_nid_t stac9205_adc_nids[2] = { - 0x12, 0x13 -}; - -static const hda_nid_t stac9205_mux_nids[2] = { - 0x19, 0x1a -}; - -static const hda_nid_t stac9205_dmux_nids[1] = { - 0x1d, -}; - -static const hda_nid_t stac9205_smux_nids[1] = { - 0x21, -}; - -#define STAC9205_NUM_DMICS 2 -static const hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { - 0x17, 0x18, 0 -}; - -#define STAC9205_NUM_CAPS 2 -static const unsigned long stac9205_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_INPUT), - HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_INPUT), -}; -static const unsigned long stac9205_capsws[] = { - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1e, 3, 0, HDA_OUTPUT), -}; - -static const hda_nid_t stac9200_pin_nids[8] = { - 0x08, 0x09, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, -}; - -static const hda_nid_t stac925x_pin_nids[8] = { - 0x07, 0x08, 0x0a, 0x0b, - 0x0c, 0x0d, 0x10, 0x11, -}; - -static const hda_nid_t stac922x_pin_nids[10] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x15, 0x1b, -}; - -static const hda_nid_t stac92hd73xx_pin_nids[13] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x22, 0x23 -}; - -#define STAC92HD71BXX_NUM_PINS 13 -static const hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x00, - 0x00, 0x14, 0x18, 0x19, 0x1e, - 0x1f, 0x20, 0x27 -}; -static const hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x18, 0x19, 0x1e, - 0x1f, 0x20, 0x27 -}; - -static const hda_nid_t stac927x_pin_nids[14] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x21, 0x22, 0x23, -}; - -static const hda_nid_t stac9205_pin_nids[12] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x16, 0x17, 0x18, - 0x21, 0x22, -}; -static int stac_add_event(struct hda_codec *codec, hda_nid_t nid, - unsigned char type, int data); -static int stac_add_hp_bass_switch(struct hda_codec *codec); -static void stac92xx_auto_set_pinctl(struct hda_codec *codec, - hda_nid_t nid, int pin_type); -static int hp_bnb2011_with_dock(struct hda_codec *codec); -static int hp_blike_system(u32 subsystem_id); -static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity); - -static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +/* + * PCM hooks + */ +static void stac_playback_pcm_hook(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream, + int action) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->dinput_mux, uinfo); + if (action == HDA_GEN_PCM_ACT_OPEN && spec->stream_delay) + msleep(spec->stream_delay); } -static int stac92xx_dmux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_capture_pcm_hook(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream, + int action) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int i, idx = 0; - ucontrol->value.enumerated.item[0] = spec->cur_dmux[dmux_idx]; - return 0; -} + if (!spec->powerdown_adcs) + return; -static int stac92xx_dmux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + for (i = 0; i < spec->gen.num_all_adcs; i++) { + if (spec->gen.all_adcs[i] == hinfo->nid) { + idx = i; + break; + } + } - return snd_hda_input_mux_put(codec, spec->dinput_mux, ucontrol, - spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + msleep(40); + snd_hda_codec_write(codec, hinfo->nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + spec->active_adcs |= (1 << idx); + break; + case HDA_GEN_PCM_ACT_CLOSE: + snd_hda_codec_write(codec, hinfo->nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + spec->active_adcs &= ~(1 << idx); + break; + } } -static int stac92xx_smux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->sinput_mux, uinfo); -} +/* + * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a + * funky external mute control using GPIO pins. + */ -static int stac92xx_smux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, + unsigned int dir_mask, unsigned int data) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + unsigned int gpiostate, gpiomask, gpiodir; - ucontrol->value.enumerated.item[0] = spec->cur_smux[smux_idx]; - return 0; + snd_printdd("%s msk %x dir %x gpio %x\n", __func__, mask, dir_mask, data); + + gpiostate = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask); + + gpiomask = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_MASK, 0); + gpiomask |= mask; + + gpiodir = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DIRECTION, 0); + gpiodir |= dir_mask; + + /* Configure GPIOx as CMOS */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); + + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, gpiomask); + snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */ + + msleep(1); + + snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } -static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* hook for controlling mic-mute LED GPIO */ +static void stac_capture_led_hook(struct hda_codec *codec, bool enable) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *smux = &spec->private_smux; - unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - int err, val; - hda_nid_t nid; - - err = snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, - spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]); - if (err < 0) - return err; + bool mute = !enable; - if (spec->spdif_mute) { - if (smux_idx == 0) - nid = spec->multiout.dig_out_nid; - else - nid = codec->slave_dig_outs[smux_idx - 1]; - if (spec->cur_smux[smux_idx] == smux->num_items - 1) - val = HDA_AMP_MUTE; + if (spec->mic_mute_led_on != mute) { + spec->mic_mute_led_on = mute; + if (mute) + spec->gpio_data |= spec->mic_mute_led_gpio; else - val = 0; - /* un/mute SPDIF out */ - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, val); + spec->gpio_data &= ~spec->mic_mute_led_gpio; + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); } - return 0; } static int stac_vrefout_set(struct hda_codec *codec, @@ -702,141 +355,233 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } -static unsigned int stac92xx_vref_set(struct hda_codec *codec, - hda_nid_t nid, unsigned int new_vref) -{ - int error; - unsigned int pincfg; - pincfg = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - - pincfg &= 0xff; - pincfg &= ~(AC_PINCTL_VREFEN | AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - pincfg |= new_vref; - - if (new_vref == AC_PINCTL_VREF_HIZ) - pincfg |= AC_PINCTL_OUT_EN; - else - pincfg |= AC_PINCTL_IN_EN; - - error = snd_hda_set_pin_ctl_cache(codec, nid, pincfg); - if (error < 0) - return error; - else - return 1; -} - -static unsigned int stac92xx_vref_get(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int vref; - vref = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - vref &= AC_PINCTL_VREFEN; - return vref; -} - -static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +/* update mute-LED accoring to the master switch */ +static void stac_update_led_status(struct hda_codec *codec, int enabled) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} + int muted = !enabled; -static int stac92xx_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + if (!spec->gpio_led) + return; - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} + /* LED state is inverted on these systems */ + if (spec->gpio_led_polarity) + muted = !muted; -static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - const struct hda_input_mux *imux = spec->input_mux; - unsigned int idx, prev_idx, didx; - - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - prev_idx = spec->cur_mux[adc_idx]; - if (prev_idx == idx) - return 0; - if (idx < spec->num_analog_muxes) { - snd_hda_codec_write_cache(codec, spec->mux_nids[adc_idx], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[idx].index); - if (prev_idx >= spec->num_analog_muxes && - spec->mux_nids[adc_idx] != spec->dmux_nids[adc_idx]) { - imux = spec->dinput_mux; - /* 0 = analog */ - snd_hda_codec_write_cache(codec, - spec->dmux_nids[adc_idx], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[0].index); - } + if (!spec->vref_mute_led_nid) { + if (muted) + spec->gpio_data |= spec->gpio_led; + else + spec->gpio_data &= ~spec->gpio_led; + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); } else { - imux = spec->dinput_mux; - /* first dimux item is hardcoded to select analog imux, - * so lets skip it - */ - didx = idx - spec->num_analog_muxes + 1; - snd_hda_codec_write_cache(codec, spec->dmux_nids[adc_idx], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[didx].index); + spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; + stac_vrefout_set(codec, spec->vref_mute_led_nid, + spec->vref_led); } - spec->cur_mux[adc_idx] = idx; - return 1; } -static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +/* vmaster hook to update mute LED */ +static void stac_vmaster_hook(void *private_data, int val) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->mono_mux, uinfo); + stac_update_led_status(private_data, val); } -static int stac92xx_mono_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +/* automute hook to handle GPIO mute and EAPD updates */ +static void stac_update_outputs(struct hda_codec *codec) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = spec->cur_mmux; - return 0; -} + if (spec->gpio_mute) + spec->gen.master_mute = + !(snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute); -static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; + snd_hda_gen_update_outputs(codec); - return snd_hda_input_mux_put(codec, spec->mono_mux, ucontrol, - spec->mono_nid, &spec->cur_mmux); + if (spec->eapd_mask && spec->eapd_switch) { + unsigned int val = spec->gpio_data; + if (spec->gen.speaker_muted) + val &= ~spec->eapd_mask; + else + val |= spec->eapd_mask; + if (spec->gpio_data != val) + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, + val); + } } -#define stac92xx_aloopback_info snd_ctl_boolean_mono_info - -static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, + bool enable, bool do_write) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); struct sigmatel_spec *spec = codec->spec; + unsigned int idx, val; - ucontrol->value.integer.value[0] = !!(spec->aloopback & - (spec->aloopback_mask << idx)); - return 0; + for (idx = 0; idx < spec->num_pwrs; idx++) { + if (spec->pwr_nids[idx] == nid) + break; + } + if (idx >= spec->num_pwrs) + return; + + idx = 1 << idx; + + val = spec->power_map_bits; + if (enable) + val &= ~idx; + else + val |= idx; + + /* power down unused output ports */ + if (val != spec->power_map_bits) { + spec->power_map_bits = val; + if (do_write) + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, val); + } +} + +/* update power bit per jack plug/unplug */ +static void jack_update_power(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + + if (!spec->num_pwrs) + return; + + if (jack && jack->nid) { + stac_toggle_power_map(codec, jack->nid, + snd_hda_jack_detect(codec, jack->nid), + true); + return; + } + + /* update all jacks */ + for (i = 0; i < spec->num_pwrs; i++) { + hda_nid_t nid = spec->pwr_nids[i]; + jack = snd_hda_jack_tbl_get(codec, nid); + if (!jack || !jack->action) + continue; + if (jack->action == STAC_PWR_EVENT || + jack->action <= HDA_GEN_LAST_EVENT) + stac_toggle_power_map(codec, nid, + snd_hda_jack_detect(codec, nid), + false); + } + + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_IDT_SET_POWER_MAP, + spec->power_map_bits); +} + +static void stac_hp_automute(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + snd_hda_gen_hp_automute(codec, jack); + jack_update_power(codec, jack); +} + +static void stac_line_automute(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + snd_hda_gen_line_automute(codec, jack); + jack_update_power(codec, jack); +} + +static void stac_vref_event(struct hda_codec *codec, struct hda_jack_tbl *event) +{ + unsigned int data; + + data = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + /* toggle VREF state based on GPIOx status */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, + !!(data & (1 << event->private_data))); +} + +/* initialize the power map and enable the power event to jacks that + * haven't been assigned to automute + */ +static void stac_init_power_map(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pwrs; i++) { + hda_nid_t nid = spec->pwr_nids[i]; + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + def_conf = get_defcfg_connect(def_conf); + if (snd_hda_jack_tbl_get(codec, nid)) + continue; + if (def_conf == AC_JACK_PORT_COMPLEX && + !(spec->vref_mute_led_nid == nid || + is_jack_detectable(codec, nid))) { + snd_hda_jack_detect_enable_callback(codec, nid, + STAC_PWR_EVENT, + jack_update_power); + } else { + if (def_conf == AC_JACK_PORT_NONE) + stac_toggle_power_map(codec, nid, false, false); + else + stac_toggle_power_map(codec, nid, true, false); + } + } +} + +/* + */ + +static inline bool get_int_hint(struct hda_codec *codec, const char *key, + int *valp) +{ + return !snd_hda_get_int_hint(codec, key, valp); +} + +/* override some hints from the hwdep entry */ +static void stac_store_hints(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int val; + + if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) { + spec->eapd_mask = spec->gpio_dir = spec->gpio_data = + spec->gpio_mask; + } + if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) + spec->gpio_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) + spec->eapd_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) + spec->gpio_mute &= spec->gpio_mask; + val = snd_hda_get_bool_hint(codec, "eapd_switch"); + if (val >= 0) + spec->eapd_switch = val; +} + +/* + * loopback controls + */ + +#define stac_aloopback_info snd_ctl_boolean_mono_info + +static int stac_aloopback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = !!(spec->aloopback & + (spec->aloopback_mask << idx)); + return 0; } -static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int stac_aloopback_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; @@ -875,399 +620,466 @@ static int stac92xx_aloopback_put(struct snd_kcontrol *kcontrol, return 1; } -static const struct hda_verb stac9200_core_init[] = { - /* set dac0mux for dac converter */ - { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {} -}; - -static const struct hda_verb stac9200_eapd_init[] = { - /* set dac0mux for dac converter */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {} -}; - -static const struct hda_verb dell_eq_core_init[] = { - /* set master volume to max value without distortion - * and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, - {} -}; - -static const struct hda_verb stac92hd73xx_core_init[] = { - /* set master volume and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - {} -}; - -static const struct hda_verb stac92hd83xxx_core_init[] = { - /* power state controls amps */ - { 0x01, AC_VERB_SET_EAPD, 1 << 2}, - {} -}; - -static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = { - { 0x22, 0x785, 0x43 }, - { 0x22, 0x782, 0xe0 }, - { 0x22, 0x795, 0x00 }, - {} -}; - -static const struct hda_verb stac92hd71bxx_core_init[] = { - /* set master volume and direct control */ - { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - {} -}; - -static const struct hda_verb stac92hd71bxx_unmute_core_init[] = { - /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {} -}; - -static const struct hda_verb stac925x_core_init[] = { - /* set dac0mux for dac converter */ - { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* mute the master volume */ - { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - {} -}; - -static const struct hda_verb stac922x_core_init[] = { - /* set master volume and direct control */ - { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - {} -}; - -static const struct hda_verb d965_core_init[] = { - /* unmute node 0x1b */ - { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* select node 0x03 as DAC */ - { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, - {} -}; - -static const struct hda_verb dell_3st_core_init[] = { - /* don't set delta bit */ - {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, - /* unmute node 0x1b */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* select node 0x03 as DAC */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, - {} -}; - -static const struct hda_verb stac927x_core_init[] = { - /* set master volume and direct control */ - { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* enable analog pc beep path */ - { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, - {} -}; - -static const struct hda_verb stac927x_volknob_core_init[] = { - /* don't set delta bit */ - {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, - /* enable analog pc beep path */ - {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, - {} -}; - -static const struct hda_verb stac9205_core_init[] = { - /* set master volume and direct control */ - { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* enable analog pc beep path */ - { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, - {} -}; - -#define STAC_MONO_MUX \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Mono Mux", \ - .count = 1, \ - .info = stac92xx_mono_mux_enum_info, \ - .get = stac92xx_mono_mux_enum_get, \ - .put = stac92xx_mono_mux_enum_put, \ - } - #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Analog Loopback", \ .count = cnt, \ - .info = stac92xx_aloopback_info, \ - .get = stac92xx_aloopback_get, \ - .put = stac92xx_aloopback_put, \ + .info = stac_aloopback_info, \ + .get = stac_aloopback_get, \ + .put = stac_aloopback_put, \ .private_value = verb_read | (verb_write << 16), \ } -#define DC_BIAS(xname, idx, nid) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = idx, \ - .info = stac92xx_dc_bias_info, \ - .get = stac92xx_dc_bias_get, \ - .put = stac92xx_dc_bias_put, \ - .private_value = nid, \ - } +/* + * Mute LED handling on HP laptops + */ -static const struct snd_kcontrol_new stac9200_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xb, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0xb, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), - { } /* end */ -}; +/* check whether it's a HP laptop with a docking port */ +static bool hp_bnb2011_with_dock(struct hda_codec *codec) +{ + if (codec->vendor_id != 0x111d7605 && + codec->vendor_id != 0x111d76d1) + return false; -static const struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - {} -}; + switch (codec->subsystem_id) { + case 0x103c1618: + case 0x103c1619: + case 0x103c161a: + case 0x103c161b: + case 0x103c161c: + case 0x103c161d: + case 0x103c161e: + case 0x103c161f: -static const struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), - {} -}; + case 0x103c162a: + case 0x103c162b: -static const struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - {} -}; + case 0x103c1630: + case 0x103c1631: + case 0x103c1633: + case 0x103c1634: + case 0x103c1635: -static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2) -}; + case 0x103c3587: + case 0x103c3588: + case 0x103c3589: + case 0x103c358a: -static const struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x0e, 0, HDA_OUTPUT), - { } /* end */ -}; + case 0x103c3667: + case 0x103c3668: + case 0x103c3669: -static const struct snd_kcontrol_new stac9205_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), - {} -}; - -static const struct snd_kcontrol_new stac927x_loopback[] = { - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), - {} -}; - -static struct snd_kcontrol_new stac_dmux_mixer = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Source", - /* count set later */ - .info = stac92xx_dmux_enum_info, - .get = stac92xx_dmux_enum_get, - .put = stac92xx_dmux_enum_put, -}; + return true; + } + return false; +} -static struct snd_kcontrol_new stac_smux_mixer = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Source", - /* count set later */ - .info = stac92xx_smux_enum_info, - .get = stac92xx_smux_enum_get, - .put = stac92xx_smux_enum_put, -}; +static bool hp_blike_system(u32 subsystem_id) +{ + switch (subsystem_id) { + case 0x103c1520: + case 0x103c1521: + case 0x103c1523: + case 0x103c1524: + case 0x103c1525: + case 0x103c1722: + case 0x103c1723: + case 0x103c1724: + case 0x103c1725: + case 0x103c1726: + case 0x103c1727: + case 0x103c1728: + case 0x103c1729: + case 0x103c172a: + case 0x103c172b: + case 0x103c307e: + case 0x103c307f: + case 0x103c3080: + case 0x103c3081: + case 0x103c7007: + case 0x103c7008: + return true; + } + return false; +} -static const char * const slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "Bass Speaker", "IEC958", "PCM", - NULL -}; +static void set_hp_led_gpio(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int gpio; -static void stac92xx_update_led_status(struct hda_codec *codec, int enabled); + if (spec->gpio_led) + return; -static void stac92xx_vmaster_hook(void *private_data, int val) -{ - stac92xx_update_led_status(private_data, val); + gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); + gpio &= AC_GPIO_IO_COUNT; + if (gpio > 3) + spec->gpio_led = 0x08; /* GPIO 3 */ + else + spec->gpio_led = 0x01; /* GPIO 0 */ } -static void stac92xx_free_kctls(struct hda_codec *codec); - -static int stac92xx_build_controls(struct hda_codec *codec) +/* + * This method searches for the mute LED GPIO configuration + * provided as OEM string in SMBIOS. The format of that string + * is HP_Mute_LED_P_G or HP_Mute_LED_P + * where P can be 0 or 1 and defines mute LED GPIO control state (low/high) + * that corresponds to the NOT muted state of the master volume + * and G is the index of the GPIO to use as the mute LED control (0..9) + * If _G portion is missing it is assigned based on the codec ID + * + * So, HP B-series like systems may have HP_Mute_LED_0 (current models) + * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + * + * + * The dv-series laptops don't seem to have the HP_Mute_LED* strings in + * SMBIOS - at least the ones I have seen do not have them - which include + * my own system (HP Pavilion dv6-1110ax) and my cousin's + * HP Pavilion dv9500t CTO. + * Need more information on whether it is true across the entire series. + * -- kunal + */ +static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) { struct sigmatel_spec *spec = codec->spec; - unsigned int vmaster_tlv[4]; - int err; - int i; + const struct dmi_device *dev = NULL; - if (spec->mixer) { - err = snd_hda_add_new_ctls(codec, spec->mixer); - if (err < 0) - return err; + if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { + get_int_hint(codec, "gpio_led_polarity", + &spec->gpio_led_polarity); + return 1; } - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - if (!spec->auto_mic && spec->num_dmuxes > 0 && - snd_hda_get_bool_hint(codec, "separate_dmux") == 1) { - stac_dmux_mixer.count = spec->num_dmuxes; - err = snd_hda_ctl_add(codec, 0, - snd_ctl_new1(&stac_dmux_mixer, codec)); - if (err < 0) - return err; - } - if (spec->num_smuxes > 0) { - int wcaps = get_wcaps(codec, spec->multiout.dig_out_nid); - struct hda_input_mux *smux = &spec->private_smux; - /* check for mute support on SPDIF out */ - if (wcaps & AC_WCAP_OUT_AMP) { - snd_hda_add_imux_item(smux, "Off", 0, NULL); - spec->spdif_mute = 1; + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { + if (sscanf(dev->name, "HP_Mute_LED_%d_%x", + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { + unsigned int max_gpio; + max_gpio = snd_hda_param_read(codec, codec->afg, + AC_PAR_GPIO_CAP); + max_gpio &= AC_GPIO_IO_COUNT; + if (spec->gpio_led < max_gpio) + spec->gpio_led = 1 << spec->gpio_led; + else + spec->vref_mute_led_nid = spec->gpio_led; + return 1; + } + if (sscanf(dev->name, "HP_Mute_LED_%d", + &spec->gpio_led_polarity) == 1) { + set_hp_led_gpio(codec); + return 1; + } + /* BIOS bug: unfilled OEM string */ + if (strstr(dev->name, "HP_Mute_LED_P_G")) { + set_hp_led_gpio(codec); + if (default_polarity >= 0) + spec->gpio_led_polarity = default_polarity; + else + spec->gpio_led_polarity = 1; + return 1; } - stac_smux_mixer.count = spec->num_smuxes; - err = snd_hda_ctl_add(codec, 0, - snd_ctl_new1(&stac_smux_mixer, codec)); - if (err < 0) - return err; } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_dig_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid, - spec->autocfg.dig_out_type[0]); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) { - err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; + /* + * Fallback case - if we don't find the DMI strings, + * we statically set the GPIO - if not a B-series system + * and default polarity is provided + */ + if (!hp_blike_system(codec->subsystem_id) && + (default_polarity == 0 || default_polarity == 1)) { + set_hp_led_gpio(codec); + spec->gpio_led_polarity = default_polarity; + return 1; } + return 0; +} - /* if we have no master control, let's create it */ - snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], - HDA_OUTPUT, vmaster_tlv); - /* correct volume offset */ - vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; - /* minimum value is actually mute */ - vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); - if (err < 0) - return err; +/* + * PC beep controls + */ - err = __snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch", true, - &spec->vmaster_mute.sw_kctl); - if (err < 0) - return err; +/* create PC beep volume controls */ +static int stac_auto_create_beep_ctls(struct hda_codec *codec, + hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); + struct snd_kcontrol_new *knew; + static struct snd_kcontrol_new abeep_mute_ctl = + HDA_CODEC_MUTE(NULL, 0, 0, 0); + static struct snd_kcontrol_new dbeep_mute_ctl = + HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0); + static struct snd_kcontrol_new beep_vol_ctl = + HDA_CODEC_VOLUME(NULL, 0, 0, 0); - if (spec->gpio_led) { - spec->vmaster_mute.hook = stac92xx_vmaster_hook; - err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); - if (err < 0) - return err; + /* check for mute support for the the amp */ + if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { + const struct snd_kcontrol_new *temp; + if (spec->anabeep_nid == nid) + temp = &abeep_mute_ctl; + else + temp = &dbeep_mute_ctl; + knew = snd_hda_gen_add_kctl(&spec->gen, + "Beep Playback Switch", temp); + if (!knew) + return -ENOMEM; + knew->private_value = + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT); } - if (spec->aloopback_ctl && - snd_hda_get_bool_hint(codec, "loopback") == 1) { - err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl); - if (err < 0) - return err; + /* check to see if there is volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + knew = snd_hda_gen_add_kctl(&spec->gen, + "Beep Playback Volume", + &beep_vol_ctl); + if (!knew) + return -ENOMEM; + knew->private_value = + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT); } + return 0; +} - stac92xx_free_kctls(codec); /* no longer needed */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP +#define stac_dig_beep_switch_info snd_ctl_boolean_mono_info - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); - if (err < 0) - return err; +static int stac_dig_beep_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = codec->beep->enabled; + return 0; +} - return 0; +static int stac_dig_beep_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]); } -static const struct hda_pintbl ref9200_pin_configs[] = { - { 0x08, 0x01c47010 }, - { 0x09, 0x01447010 }, - { 0x0d, 0x0221401f }, - { 0x0e, 0x01114010 }, - { 0x0f, 0x02a19020 }, - { 0x10, 0x01a19021 }, - { 0x11, 0x90100140 }, - { 0x12, 0x01813122 }, - {} +static const struct snd_kcontrol_new stac_dig_beep_ctrl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Beep Playback Switch", + .info = stac_dig_beep_switch_info, + .get = stac_dig_beep_switch_get, + .put = stac_dig_beep_switch_put, }; -static const struct hda_pintbl gateway9200_m4_pin_configs[] = { - { 0x08, 0x400000fe }, - { 0x09, 0x404500f4 }, - { 0x0d, 0x400100f0 }, - { 0x0e, 0x90110010 }, - { 0x0f, 0x400100f1 }, - { 0x10, 0x02a1902e }, - { 0x11, 0x500000f2 }, - { 0x12, 0x500000f3 }, +static int stac_beep_switch_ctl(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &stac_dig_beep_ctrl)) + return -ENOMEM; + return 0; +} +#endif + +/* + */ + +static const struct hda_verb stac9200_core_init[] = { + /* set dac0mux for dac converter */ + { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {} }; -static const struct hda_pintbl gateway9200_m4_2_pin_configs[] = { - { 0x08, 0x400000fe }, - { 0x09, 0x404500f4 }, - { 0x0d, 0x400100f0 }, - { 0x0e, 0x90110010 }, - { 0x0f, 0x400100f1 }, - { 0x10, 0x02a1902e }, - { 0x11, 0x500000f2 }, - { 0x12, 0x500000f3 }, +static const struct hda_verb stac9200_eapd_init[] = { + /* set dac0mux for dac converter */ + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {} }; -/* - STAC 9200 pin configs for - 102801A8 - 102801DE - 102801E8 -*/ -static const struct hda_pintbl dell9200_d21_pin_configs[] = { - { 0x08, 0x400001f0 }, - { 0x09, 0x400001f1 }, - { 0x0d, 0x02214030 }, - { 0x0e, 0x01014010 }, - { 0x0f, 0x02a19020 }, - { 0x10, 0x01a19021 }, - { 0x11, 0x90100140 }, - { 0x12, 0x01813122 }, +static const struct hda_verb dell_eq_core_init[] = { + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, {} }; -/* - STAC 9200 pin configs for - 102801C0 - 102801C1 -*/ -static const struct hda_pintbl dell9200_d22_pin_configs[] = { - { 0x08, 0x400001f0 }, - { 0x09, 0x400001f1 }, - { 0x0d, 0x0221401f }, - { 0x0e, 0x01014010 }, - { 0x0f, 0x01813020 }, - { 0x10, 0x02a19021 }, - { 0x11, 0x90100140 }, - { 0x12, 0x400001f2 }, +static const struct hda_verb stac92hd73xx_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, {} }; -/* +static const struct hda_verb stac92hd83xxx_core_init[] = { + /* power state controls amps */ + { 0x01, AC_VERB_SET_EAPD, 1 << 2}, + {} +}; + +static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = { + { 0x22, 0x785, 0x43 }, + { 0x22, 0x782, 0xe0 }, + { 0x22, 0x795, 0x00 }, + {} +}; + +static const struct hda_verb stac92hd71bxx_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + {} +}; + +static const struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} +}; + +static const struct hda_verb stac925x_core_init[] = { + /* set dac0mux for dac converter */ + { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* mute the master volume */ + { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + {} +}; + +static const struct hda_verb stac922x_core_init[] = { + /* set master volume and direct control */ + { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + {} +}; + +static const struct hda_verb d965_core_init[] = { + /* unmute node 0x1b */ + { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {} +}; + +static const struct hda_verb dell_3st_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* unmute node 0x1b */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {} +}; + +static const struct hda_verb stac927x_core_init[] = { + /* set master volume and direct control */ + { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + +static const struct hda_verb stac927x_volknob_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* enable analog pc beep path */ + {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + +static const struct hda_verb stac9205_core_init[] = { + /* set master volume and direct control */ + { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + +static const struct snd_kcontrol_new stac92hd73xx_6ch_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3); + +static const struct snd_kcontrol_new stac92hd73xx_8ch_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4); + +static const struct snd_kcontrol_new stac92hd73xx_10ch_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5); + +static const struct snd_kcontrol_new stac92hd71bxx_loopback = + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2); + +static const struct snd_kcontrol_new stac9205_loopback = + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1); + +static const struct snd_kcontrol_new stac927x_loopback = + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1); + +static const struct hda_pintbl ref9200_pin_configs[] = { + { 0x08, 0x01c47010 }, + { 0x09, 0x01447010 }, + { 0x0d, 0x0221401f }, + { 0x0e, 0x01114010 }, + { 0x0f, 0x02a19020 }, + { 0x10, 0x01a19021 }, + { 0x11, 0x90100140 }, + { 0x12, 0x01813122 }, + {} +}; + +static const struct hda_pintbl gateway9200_m4_pin_configs[] = { + { 0x08, 0x400000fe }, + { 0x09, 0x404500f4 }, + { 0x0d, 0x400100f0 }, + { 0x0e, 0x90110010 }, + { 0x0f, 0x400100f1 }, + { 0x10, 0x02a1902e }, + { 0x11, 0x500000f2 }, + { 0x12, 0x500000f3 }, + {} +}; + +static const struct hda_pintbl gateway9200_m4_2_pin_configs[] = { + { 0x08, 0x400000fe }, + { 0x09, 0x404500f4 }, + { 0x0d, 0x400100f0 }, + { 0x0e, 0x90110010 }, + { 0x0f, 0x400100f1 }, + { 0x10, 0x02a1902e }, + { 0x11, 0x500000f2 }, + { 0x12, 0x500000f3 }, + {} +}; + +/* + STAC 9200 pin configs for + 102801A8 + 102801DE + 102801E8 +*/ +static const struct hda_pintbl dell9200_d21_pin_configs[] = { + { 0x08, 0x400001f0 }, + { 0x09, 0x400001f1 }, + { 0x0d, 0x02214030 }, + { 0x0e, 0x01014010 }, + { 0x0f, 0x02a19020 }, + { 0x10, 0x01a19021 }, + { 0x11, 0x90100140 }, + { 0x12, 0x01813122 }, + {} +}; + +/* + STAC 9200 pin configs for + 102801C0 + 102801C1 +*/ +static const struct hda_pintbl dell9200_d22_pin_configs[] = { + { 0x08, 0x400001f0 }, + { 0x09, 0x400001f1 }, + { 0x0d, 0x0221401f }, + { 0x0e, 0x01014010 }, + { 0x0f, 0x01813020 }, + { 0x10, 0x02a19021 }, + { 0x11, 0x90100140 }, + { 0x12, 0x400001f2 }, + {} +}; + +/* STAC 9200 pin configs for 102801C4 (Dell Dimension E310) 102801C5 @@ -1431,17 +1243,13 @@ static void stac9200_fixup_panasonic(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gpio_mask = spec->gpio_dir = 0x09; spec->gpio_data = 0x00; - break; - case HDA_FIXUP_ACT_PROBE: /* CF-74 has no headphone detection, and the driver should *NOT* * do detection and HP/speaker toggle because the hardware does it. */ - spec->hp_detect = 0; - break; + spec->gen.suppress_auto_mute = 1; } } @@ -1862,7 +1670,6 @@ static void stac92hd73xx_fixup_dell(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; snd_hda_apply_pincfgs(codec, dell_m6_pin_configs); - spec->num_smuxes = 0; spec->eapd_switch = 0; } @@ -1883,43 +1690,34 @@ static void stac92hd73xx_fixup_dell_eq(struct hda_codec *codec, static void stac92hd73xx_fixup_dell_m6_amic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - if (action != HDA_FIXUP_ACT_PRE_PROBE) return; stac92hd73xx_fixup_dell(codec); snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); - spec->num_dmics = 0; } /* Digital Mics */ static void stac92hd73xx_fixup_dell_m6_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - if (action != HDA_FIXUP_ACT_PRE_PROBE) return; stac92hd73xx_fixup_dell(codec); snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); - spec->num_dmics = 1; } /* Both */ static void stac92hd73xx_fixup_dell_m6_both(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - if (action != HDA_FIXUP_ACT_PRE_PROBE) return; stac92hd73xx_fixup_dell(codec); snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); - spec->num_dmics = 1; } static void stac92hd73xx_fixup_alienware_m17x(struct hda_codec *codec, @@ -1931,19 +1729,14 @@ static void stac92hd73xx_fixup_alienware_m17x(struct hda_codec *codec, return; snd_hda_apply_pincfgs(codec, alienware_m17x_pin_configs); - spec->num_dmics = STAC92HD73XX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); spec->eapd_switch = 0; } static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PROBE) - return; - spec->hp_detect = 0; + if (action == HDA_FIXUP_ACT_PRE_PROBE) + codec->no_jack_detect = 1; } static const struct hda_fixup stac92hd73xx_fixups[] = { @@ -2348,6 +2141,54 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { {} /* terminator */ }; +/* HP dv7 bass switch - GPIO5 */ +#define stac_hp_bass_gpio_info snd_ctl_boolean_mono_info +static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20); + return 0; +} + +static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_data; + + gpio_data = (spec->gpio_data & ~0x20) | + (ucontrol->value.integer.value[0] ? 0x20 : 0); + if (gpio_data == spec->gpio_data) + return 0; + spec->gpio_data = gpio_data; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); + return 1; +} + +static const struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = stac_hp_bass_gpio_info, + .get = stac_hp_bass_gpio_get, + .put = stac_hp_bass_gpio_put, +}; + +static int stac_add_hp_bass_switch(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!snd_hda_gen_add_kctl(&spec->gen, "Bass Speaker Playback Switch", + &stac_hp_bass_sw_ctrl)) + return -ENOMEM; + + spec->gpio_mask |= 0x20; + spec->gpio_dir |= 0x20; + spec->gpio_data |= 0x20; + return 0; +} + static const struct hda_pintbl ref92hd71bxx_pin_configs[] = { { 0x0a, 0x02214030 }, { 0x0b, 0x02a19040 }, @@ -2420,66 +2261,29 @@ static void stac92hd71bxx_fixup_ref(struct hda_codec *codec, spec->gpio_mask = spec->gpio_dir = spec->gpio_data = 0; } -static void stac92hd71bxx_fixup_no_dmic(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - spec->num_dmics = 0; - spec->num_smuxes = 0; - spec->num_dmuxes = 0; -} - -static void stac92hd71bxx_fixup_no_smux(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - spec->num_dmics = 1; - spec->num_smuxes = 0; - spec->num_dmuxes = 1; -} - static void stac92hd71bxx_fixup_hp_m4(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct sigmatel_spec *spec = codec->spec; + struct hda_jack_tbl *jack; if (action != HDA_FIXUP_ACT_PRE_PROBE) return; /* Enable VREF power saving on GPIO1 detect */ - stac_add_event(codec, codec->afg, STAC_VREF_EVENT, 0x02); snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); - snd_hda_jack_detect_enable(codec, codec->afg, 0); + snd_hda_jack_detect_enable_callback(codec, codec->afg, + STAC_VREF_EVENT, + stac_vref_event); + jack = snd_hda_jack_tbl_get(codec, codec->afg); + if (jack) + jack->private_data = 0x02; + spec->gpio_mask |= 0x02; /* enable internal microphone */ snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); - stac92xx_auto_set_pinctl(codec, 0x0e, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); - - stac92hd71bxx_fixup_no_dmic(codec, fix, action); -} - -static void stac92hd71bxx_fixup_hp_dv4_1222nr(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - struct sigmatel_spec *spec = codec->spec; - - if (action != HDA_FIXUP_ACT_PRE_PROBE) - return; - spec->num_dmics = 1; - /* I don't know if it needs 1 or 2 smuxes - will wait for - * bug reports to fix if needed - */ - spec->num_smuxes = 1; - spec->num_dmuxes = 1; } static void stac92hd71bxx_fixup_hp_dv4(struct hda_codec *codec, @@ -2495,21 +2299,14 @@ static void stac92hd71bxx_fixup_hp_dv4(struct hda_codec *codec, static void stac92hd71bxx_fixup_hp_dv5(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; unsigned int cap; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); - stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); break; case HDA_FIXUP_ACT_PROBE: - /* HP dv6 gives the headphone pin as a line-out. Thus we - * need to set hp_detect flag here to force to enable HP - * detection. - */ - spec->hp_detect = 1; /* enable bass on HP dv7 */ cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP); cap &= AC_GPIO_IO_COUNT; @@ -2527,9 +2324,6 @@ static void stac92hd71bxx_fixup_hp_hdx(struct hda_codec *codec, if (action != HDA_FIXUP_ACT_PRE_PROBE) return; spec->gpio_led = 0x08; - spec->num_dmics = 1; - spec->num_smuxes = 1; - spec->num_dmuxes = 1; } @@ -2559,7 +2353,7 @@ static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, } } - if (find_mute_led_cfg(codec, spec->default_polarity)) + if (find_mute_led_cfg(codec, 1)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, spec->gpio_led_polarity); @@ -2574,20 +2368,14 @@ static const struct hda_fixup stac92hd71bxx_fixups[] = { [STAC_DELL_M4_1] = { .type = HDA_FIXUP_PINS, .v.pins = dell_m4_1_pin_configs, - .chained = true, - .chain_id = STAC_92HD71BXX_NO_SMUX, }, [STAC_DELL_M4_2] = { .type = HDA_FIXUP_PINS, .v.pins = dell_m4_2_pin_configs, - .chained = true, - .chain_id = STAC_92HD71BXX_NO_DMIC, }, [STAC_DELL_M4_3] = { .type = HDA_FIXUP_PINS, .v.pins = dell_m4_3_pin_configs, - .chained = true, - .chain_id = STAC_92HD71BXX_NO_SMUX, }, [STAC_HP_M4] = { .type = HDA_FIXUP_FUNC, @@ -2613,23 +2401,9 @@ static const struct hda_fixup stac92hd71bxx_fixups[] = { .chained = true, .chain_id = STAC_92HD71BXX_HP, }, - [STAC_HP_DV4_1222NR] = { + [STAC_92HD71BXX_HP] = { .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_hp_dv4_1222nr, - .chained = true, - .chain_id = STAC_HP_DV4, - }, - [STAC_92HD71BXX_NO_DMIC] = { - .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_no_dmic, - }, - [STAC_92HD71BXX_NO_SMUX] = { - .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_no_smux, - }, - [STAC_92HD71BXX_HP] = { - .type = HDA_FIXUP_FUNC, - .v.func = stac92hd71bxx_fixup_hp, + .v.func = stac92hd71bxx_fixup_hp, }, }; @@ -2642,7 +2416,7 @@ static const struct hda_model_fixup stac92hd71bxx_models[] = { { .id = STAC_HP_DV4, .name = "hp-dv4" }, { .id = STAC_HP_DV5, .name = "hp-dv5" }, { .id = STAC_HP_HDX, .name = "hp-hdx" }, - { .id = STAC_HP_DV4_1222NR, .name = "hp-dv4-1222nr" }, + { .id = STAC_HP_DV4, .name = "hp-dv4-1222nr" }, {} }; @@ -2652,8 +2426,6 @@ static const struct snd_pci_quirk stac92hd71bxx_fixup_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "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, @@ -3234,11 +3006,9 @@ static const struct hda_pintbl dell_3st_pin_configs[] = { static void stac927x_fixup_ref_no_jd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct sigmatel_spec *spec = codec->spec; - /* no jack detecion for ref-no-jd model */ - if (action == HDA_FIXUP_ACT_PROBE) - spec->hp_detect = 0; + if (action == HDA_FIXUP_ACT_PRE_PROBE) + codec->no_jack_detect = 1; } static void stac927x_fixup_ref(struct hda_codec *codec, @@ -3266,13 +3036,9 @@ static void stac927x_fixup_dell_dmic(struct hda_codec *codec, spec->eapd_mask = spec->gpio_mask = 0x04; spec->gpio_dir = spec->gpio_data = 0x04; } - spec->dmic_nids = stac927x_dmic_nids; - spec->num_dmics = STAC927X_NUM_DMICS; snd_hda_add_verbs(codec, dell_3st_core_init); spec->volknob_init = 1; - spec->dmux_nids = stac927x_dmux_nids; - spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); } static void stac927x_fixup_volknob(struct hda_codec *codec, @@ -3511,18 +3277,20 @@ static void stac9205_fixup_dell_m43(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct sigmatel_spec *spec = codec->spec; - int err; + struct hda_jack_tbl *jack; if (action == HDA_FIXUP_ACT_PRE_PROBE) { snd_hda_apply_pincfgs(codec, dell_9205_m43_pin_configs); /* Enable unsol response for GPIO4/Dock HP connection */ - err = stac_add_event(codec, codec->afg, STAC_VREF_EVENT, 0x01); - if (err < 0) - return; snd_hda_codec_write_cache(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); - snd_hda_jack_detect_enable(codec, codec->afg, 0); + snd_hda_jack_detect_enable_callback(codec, codec->afg, + STAC_VREF_EVENT, + stac_vref_event); + jack = snd_hda_jack_tbl_get(codec, codec->afg); + if (jack) + jack->private_data = 0x01; spec->gpio_dir = 0x0b; spec->eapd_mask = 0x01; @@ -3610,2687 +3378,163 @@ static const struct snd_pci_quirk stac9205_fixup_tbl[] = { "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, - "Dell Precision", STAC_9205_DELL_M43), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, - "Dell Inspiron", STAC_9205_DELL_M44), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, - "Dell Vostro 1500", STAC_9205_DELL_M42), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229, - "Dell Vostro 1700", STAC_9205_DELL_M42), - /* Gateway */ - SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD), - SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), - {} /* terminator */ -}; - -/* - * Analog playback callbacks - */ -static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - if (spec->stream_delay) - msleep(spec->stream_delay); - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int stac92xx_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, format, substream); -} - -static int stac92xx_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital playback callbacks - */ -static int stac92xx_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int stac92xx_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); -} - - -/* - * Analog capture callbacks - */ -static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = spec->adc_nids[substream->number]; - - if (spec->powerdown_adcs) { - msleep(40); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - } - snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); - return 0; -} - -static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = spec->adc_nids[substream->number]; - - snd_hda_codec_cleanup_stream(codec, nid); - if (spec->powerdown_adcs) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - return 0; -} - -static const struct hda_pcm_stream stac92xx_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in stac92xx_build_pcms */ - .ops = { - .open = stac92xx_dig_playback_pcm_open, - .close = stac92xx_dig_playback_pcm_close, - .prepare = stac92xx_dig_playback_pcm_prepare, - .cleanup = stac92xx_dig_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream stac92xx_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in stac92xx_build_pcms */ -}; - -static const struct hda_pcm_stream stac92xx_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 8, - .nid = 0x02, /* NID to query formats and rates */ - .ops = { - .open = stac92xx_playback_pcm_open, - .prepare = stac92xx_playback_pcm_prepare, - .cleanup = stac92xx_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream stac92xx_pcm_analog_alt_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0x06, /* NID to query formats and rates */ - .ops = { - .open = stac92xx_playback_pcm_open, - .prepare = stac92xx_playback_pcm_prepare, - .cleanup = stac92xx_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream stac92xx_pcm_analog_capture = { - .channels_min = 2, - .channels_max = 2, - /* NID + .substreams is set in stac92xx_build_pcms */ - .ops = { - .prepare = stac92xx_capture_pcm_prepare, - .cleanup = stac92xx_capture_pcm_cleanup - }, -}; - -static int stac92xx_build_pcms(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "STAC92xx Analog"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->multiout.dac_nids[0]; - if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && - spec->autocfg.line_outs == 2) - info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap = - snd_pcm_2_1_chmaps; - - info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs; - - if (spec->alt_switch) { - codec->num_pcms++; - info++; - info->name = "STAC92xx Analog Alt"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_alt_playback; - } - - if (spec->multiout.dig_out_nid || spec->dig_in_nid) { - codec->num_pcms++; - info++; - info->name = "STAC92xx Digital"; - info->pcm_type = spec->autocfg.dig_out_type[0]; - if (spec->multiout.dig_out_nid) { - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - } - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } - } - - return 0; -} - -static void stac92xx_auto_set_pinctl(struct hda_codec *codec, hda_nid_t nid, int pin_type) - -{ - snd_hda_set_pin_ctl_cache(codec, nid, pin_type); -} - -#define stac92xx_hp_switch_info snd_ctl_boolean_mono_info - -static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.integer.value[0] = !!spec->hp_switch; - return 0; -} - -static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid); - -static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - int nid = kcontrol->private_value; - - spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0; - - /* check to be sure that the ports are up to date with - * switch changes - */ - stac_issue_unsol_event(codec, nid); - - return 1; -} - -static int stac92xx_dc_bias_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int i; - static const char * const texts[] = { - "Mic In", "Line In", "Line Out" - }; - - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value; - - if (nid == spec->mic_switch || nid == spec->line_switch) - i = 3; - else - i = 2; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = i; - uinfo->count = 1; - if (uinfo->value.enumerated.item >= i) - uinfo->value.enumerated.item = i-1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - -static int stac92xx_dc_bias_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value; - unsigned int vref = stac92xx_vref_get(codec, nid); - - if (vref == snd_hda_get_default_vref(codec, nid)) - ucontrol->value.enumerated.item[0] = 0; - else if (vref == AC_PINCTL_VREF_GRD) - ucontrol->value.enumerated.item[0] = 1; - else if (vref == AC_PINCTL_VREF_HIZ) - ucontrol->value.enumerated.item[0] = 2; - - return 0; -} - -static int stac92xx_dc_bias_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int new_vref = 0; - int error; - hda_nid_t nid = kcontrol->private_value; - - if (ucontrol->value.enumerated.item[0] == 0) - new_vref = snd_hda_get_default_vref(codec, nid); - else if (ucontrol->value.enumerated.item[0] == 1) - new_vref = AC_PINCTL_VREF_GRD; - else if (ucontrol->value.enumerated.item[0] == 2) - new_vref = AC_PINCTL_VREF_HIZ; - else - return 0; - - if (new_vref != stac92xx_vref_get(codec, nid)) { - error = stac92xx_vref_set(codec, nid, new_vref); - return error; - } - - return 0; -} - -static int stac92xx_io_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - char *texts[2]; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - if (kcontrol->private_value == spec->line_switch) - texts[0] = "Line In"; - else - texts[0] = "Mic In"; - texts[1] = "Line Out"; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->value.enumerated.items = 2; - uinfo->count = 1; - - if (uinfo->value.enumerated.item >= 2) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - - return 0; -} - -static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value; - int io_idx = (nid == spec->mic_switch) ? 1 : 0; - - ucontrol->value.enumerated.item[0] = spec->io_switch[io_idx]; - return 0; -} - -static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value; - int io_idx = (nid == spec->mic_switch) ? 1 : 0; - unsigned short val = !!ucontrol->value.enumerated.item[0]; - - spec->io_switch[io_idx] = val; - - if (val) - stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); - else { - unsigned int pinctl = AC_PINCTL_IN_EN; - if (io_idx) /* set VREF for mic */ - pinctl |= snd_hda_get_default_vref(codec, nid); - stac92xx_auto_set_pinctl(codec, nid, pinctl); - } - - /* check the auto-mute again: we need to mute/unmute the speaker - * appropriately according to the pin direction - */ - if (spec->hp_detect) - stac_issue_unsol_event(codec, nid); - - return 1; -} - -#define stac92xx_clfe_switch_info snd_ctl_boolean_mono_info - -static int stac92xx_clfe_switch_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.integer.value[0] = spec->clfe_swap; - return 0; -} - -static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value & 0xff; - unsigned int val = !!ucontrol->value.integer.value[0]; - - if (spec->clfe_swap == val) - return 0; - - spec->clfe_swap = val; - - snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, - spec->clfe_swap ? 0x4 : 0x0); - - return 1; -} - -#define STAC_CODEC_HP_SWITCH(xname) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .info = stac92xx_hp_switch_info, \ - .get = stac92xx_hp_switch_get, \ - .put = stac92xx_hp_switch_put, \ - } - -#define STAC_CODEC_IO_SWITCH(xname, xpval) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .info = stac92xx_io_switch_info, \ - .get = stac92xx_io_switch_get, \ - .put = stac92xx_io_switch_put, \ - .private_value = xpval, \ - } - -#define STAC_CODEC_CLFE_SWITCH(xname, xpval) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .info = stac92xx_clfe_switch_info, \ - .get = stac92xx_clfe_switch_get, \ - .put = stac92xx_clfe_switch_put, \ - .private_value = xpval, \ - } - -enum { - STAC_CTL_WIDGET_VOL, - STAC_CTL_WIDGET_MUTE, - STAC_CTL_WIDGET_MUTE_BEEP, - STAC_CTL_WIDGET_MONO_MUX, - STAC_CTL_WIDGET_HP_SWITCH, - STAC_CTL_WIDGET_IO_SWITCH, - STAC_CTL_WIDGET_CLFE_SWITCH, - STAC_CTL_WIDGET_DC_BIAS -}; - -static const struct snd_kcontrol_new stac92xx_control_templates[] = { - HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), - HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0), - STAC_MONO_MUX, - STAC_CODEC_HP_SWITCH(NULL), - STAC_CODEC_IO_SWITCH(NULL, 0), - STAC_CODEC_CLFE_SWITCH(NULL, 0), - DC_BIAS(NULL, 0, 0), -}; - -/* add dynamic controls */ -static struct snd_kcontrol_new * -stac_control_new(struct sigmatel_spec *spec, - const struct snd_kcontrol_new *ktemp, - const char *name, - unsigned int subdev) -{ - struct snd_kcontrol_new *knew; - - knew = snd_array_new(&spec->kctls); - if (!knew) - return NULL; - *knew = *ktemp; - knew->name = kstrdup(name, GFP_KERNEL); - if (!knew->name) { - /* roolback */ - memset(knew, 0, sizeof(*knew)); - spec->kctls.alloced--; - return NULL; - } - knew->subdevice = subdev; - return knew; -} - -static struct snd_kcontrol_new * -add_control_temp(struct sigmatel_spec *spec, - const struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) -{ - struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, - HDA_SUBDEV_AMP_FLAG); - if (!knew) - return NULL; - knew->index = idx; - knew->private_value = val; - return knew; -} - -static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - const struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) -{ - return add_control_temp(spec, ktemp, idx, name, val) ? 0 : -ENOMEM; -} - -static inline int stac92xx_add_control_idx(struct sigmatel_spec *spec, - int type, int idx, const char *name, - unsigned long val) -{ - return stac92xx_add_control_temp(spec, - &stac92xx_control_templates[type], - idx, name, val); -} - - -/* add dynamic controls */ -static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, - const char *name, unsigned long val) -{ - return stac92xx_add_control_idx(spec, type, 0, name, val); -} - -static const struct snd_kcontrol_new stac_input_src_temp = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Input Source", - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, -}; - -static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, - hda_nid_t nid, int idx) -{ - int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int control = 0; - struct sigmatel_spec *spec = codec->spec; - char name[22]; - - if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { - if (spec->headset_jack && snd_hda_get_input_pin_attr(def_conf) - != INPUT_PIN_ATTR_DOCK) - return 0; - if (snd_hda_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD - && nid == spec->line_switch) - control = STAC_CTL_WIDGET_IO_SWITCH; - else if (snd_hda_query_pin_caps(codec, nid) - & (AC_PINCAP_VREF_GRD << AC_PINCAP_VREF_SHIFT)) - control = STAC_CTL_WIDGET_DC_BIAS; - else if (nid == spec->mic_switch) - control = STAC_CTL_WIDGET_IO_SWITCH; - } - - if (control) { - snd_hda_get_pin_label(codec, nid, &spec->autocfg, - name, sizeof(name), NULL); - return stac92xx_add_control(codec->spec, control, - strcat(name, " Jack Mode"), nid); - } - - return 0; -} - -static int stac92xx_add_input_source(struct sigmatel_spec *spec) -{ - struct snd_kcontrol_new *knew; - struct hda_input_mux *imux = &spec->private_imux; - - if (spec->auto_mic) - return 0; /* no need for input source */ - if (!spec->num_adcs || imux->num_items <= 1) - return 0; /* no need for input source control */ - knew = stac_control_new(spec, &stac_input_src_temp, - stac_input_src_temp.name, 0); - if (!knew) - return -ENOMEM; - knew->count = spec->num_adcs; - return 0; -} - -/* check whether the line-input can be used as line-out */ -static hda_nid_t check_line_out_switch(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t nid; - unsigned int pincap; - int i; - - if (cfg->line_out_type != AUTO_PIN_LINE_OUT) - return 0; - for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].type == AUTO_PIN_LINE_IN) { - nid = cfg->inputs[i].pin; - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_OUT) - return nid; - } - } - return 0; -} - -static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid); - -/* check whether the mic-input can be used as line-out */ -static hda_nid_t check_mic_out_switch(struct hda_codec *codec, hda_nid_t *dac) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int def_conf, pincap; - int i; - - *dac = 0; - if (cfg->line_out_type != AUTO_PIN_LINE_OUT) - return 0; - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; - if (cfg->inputs[i].type != AUTO_PIN_MIC) - continue; - def_conf = snd_hda_codec_get_pincfg(codec, nid); - /* some laptops have an internal analog microphone - * which can't be used as a output */ - if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_OUT) { - *dac = get_unassigned_dac(codec, nid); - if (*dac) - return nid; - } - } - } - return 0; -} - -static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - for (i = 0; i < spec->multiout.num_dacs; i++) { - if (spec->multiout.dac_nids[i] == nid) - return 1; - } - - return 0; -} - -static int check_all_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - if (is_in_dac_nids(spec, nid)) - return 1; - for (i = 0; i < spec->autocfg.hp_outs; i++) - if (spec->hp_dacs[i] == nid) - return 1; - for (i = 0; i < spec->autocfg.speaker_outs; i++) - if (spec->speaker_dacs[i] == nid) - return 1; - return 0; -} - -static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int j, conn_len; - hda_nid_t conn[HDA_MAX_CONNECTIONS], fallback_dac; - unsigned int wcaps, wtype; - - conn_len = snd_hda_get_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); - /* 92HD88: trace back up the link of nids to find the DAC */ - while (conn_len == 1 && (get_wcaps_type(get_wcaps(codec, conn[0])) - != AC_WID_AUD_OUT)) { - nid = conn[0]; - conn_len = snd_hda_get_connections(codec, nid, conn, - HDA_MAX_CONNECTIONS); - } - for (j = 0; j < conn_len; j++) { - wcaps = get_wcaps(codec, conn[j]); - wtype = get_wcaps_type(wcaps); - /* we check only analog outputs */ - if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) - continue; - /* if this route has a free DAC, assign it */ - if (!check_all_dac_nids(spec, conn[j])) { - if (conn_len > 1) { - /* select this DAC in the pin's input mux */ - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, j); - } - return conn[j]; - } - } - - /* if all DACs are already assigned, connect to the primary DAC, - unless we're assigning a secondary headphone */ - fallback_dac = spec->multiout.dac_nids[0]; - if (spec->multiout.hp_nid) { - for (j = 0; j < cfg->hp_outs; j++) - if (cfg->hp_pins[j] == nid) { - fallback_dac = spec->multiout.hp_nid; - break; - } - } - - if (conn_len > 1) { - for (j = 0; j < conn_len; j++) { - if (conn[j] == fallback_dac) { - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, j); - break; - } - } - } - return 0; -} - -static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid); -static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid); - -/* - * Fill in the dac_nids table from the parsed pin configuration - * This function only works when every pin in line_out_pins[] - * contains atleast one DAC in its connection list. Some 92xx - * codecs are not connected directly to a DAC, such as the 9200 - * and 9202/925x. For those, dac_nids[] must be hard-coded. - */ -static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; - hda_nid_t nid, dac; - - for (i = 0; i < cfg->line_outs; i++) { - nid = cfg->line_out_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (!dac) { - if (spec->multiout.num_dacs > 0) { - /* we have already working output pins, - * so let's drop the broken ones again - */ - cfg->line_outs = spec->multiout.num_dacs; - break; - } - /* error out, no available DAC found */ - snd_printk(KERN_ERR - "%s: No available DAC for pin 0x%x\n", - __func__, nid); - return -ENODEV; - } - add_spec_dacs(spec, dac); - } - - for (i = 0; i < cfg->hp_outs; i++) { - nid = cfg->hp_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) { - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = dac; - else - add_spec_extra_dacs(spec, dac); - } - spec->hp_dacs[i] = dac; - } - - for (i = 0; i < cfg->speaker_outs; i++) { - nid = cfg->speaker_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) - add_spec_extra_dacs(spec, dac); - spec->speaker_dacs[i] = dac; - } - - /* add line-in as output */ - nid = check_line_out_switch(codec); - if (nid) { - dac = get_unassigned_dac(codec, nid); - if (dac) { - snd_printdd("STAC: Add line-in 0x%x as output %d\n", - nid, cfg->line_outs); - cfg->line_out_pins[cfg->line_outs] = nid; - cfg->line_outs++; - spec->line_switch = nid; - add_spec_dacs(spec, dac); - } - } - /* add mic as output */ - nid = check_mic_out_switch(codec, &dac); - if (nid && dac) { - snd_printdd("STAC: Add mic-in 0x%x as output %d\n", - nid, cfg->line_outs); - cfg->line_out_pins[cfg->line_outs] = nid; - cfg->line_outs++; - spec->mic_switch = nid; - add_spec_dacs(spec, dac); - } - - snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", - spec->multiout.num_dacs, - spec->multiout.dac_nids[0], - spec->multiout.dac_nids[1], - spec->multiout.dac_nids[2], - spec->multiout.dac_nids[3], - spec->multiout.dac_nids[4]); - - return 0; -} - -/* create volume control/switch for the given prefx type */ -static int create_controls_idx(struct hda_codec *codec, const char *pfx, - int idx, hda_nid_t nid, int chs) -{ - struct sigmatel_spec *spec = codec->spec; - char name[32]; - int err; - - if (!spec->check_volume_offset) { - unsigned int caps, step, nums, db_scale; - caps = query_amp_caps(codec, nid, HDA_OUTPUT); - step = (caps & AC_AMPCAP_STEP_SIZE) >> - AC_AMPCAP_STEP_SIZE_SHIFT; - step = (step + 1) * 25; /* in .01dB unit */ - nums = (caps & AC_AMPCAP_NUM_STEPS) >> - AC_AMPCAP_NUM_STEPS_SHIFT; - db_scale = nums * step; - /* if dB scale is over -64dB, and finer enough, - * let's reduce it to half - */ - if (db_scale > 6400 && nums >= 0x1f) - spec->volume_offset = nums / 2; - spec->check_volume_offset = 1; - } - - sprintf(name, "%s Playback Volume", pfx); - err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name, - HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, - spec->volume_offset)); - if (err < 0) - return err; - sprintf(name, "%s Playback Switch", pfx); - err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name, - HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); - if (err < 0) - return err; - return 0; -} - -#define create_controls(codec, pfx, nid, chs) \ - create_controls_idx(codec, pfx, 0, nid, chs) - -static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) -{ - if (spec->multiout.num_dacs > 4) { - printk(KERN_WARNING "stac92xx: No space for DAC 0x%x\n", nid); - return 1; - } else { - snd_BUG_ON(spec->multiout.dac_nids != spec->dac_nids); - spec->dac_nids[spec->multiout.num_dacs] = nid; - spec->multiout.num_dacs++; - } - return 0; -} - -static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { - if (!spec->multiout.extra_out_nid[i]) { - spec->multiout.extra_out_nid[i] = nid; - return 0; - } - } - printk(KERN_WARNING "stac92xx: No space for extra DAC 0x%x\n", nid); - return 1; -} - -/* Create output controls - * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT) - */ -static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, - const hda_nid_t *pins, - const hda_nid_t *dac_nids, - int type) -{ - struct sigmatel_spec *spec = codec->spec; - static const char * const chname[4] = { - "Front", "Surround", NULL /*CLFE*/, "Side" - }; - hda_nid_t nid; - int i, err; - unsigned int wid_caps; - - for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { - if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { - if (is_jack_detectable(codec, pins[i])) - spec->hp_detect = 1; - } - nid = dac_nids[i]; - if (!nid) - continue; - if (type != AUTO_PIN_HP_OUT && i == 2) { - /* Center/LFE */ - err = create_controls(codec, "Center", nid, 1); - if (err < 0) - return err; - err = create_controls(codec, "LFE", nid, 2); - if (err < 0) - return err; - - wid_caps = get_wcaps(codec, nid); - - if (wid_caps & AC_WCAP_LR_SWAP) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_CLFE_SWITCH, - "Swap Center/LFE Playback Switch", nid); - - if (err < 0) - return err; - } - - } else { - const char *name; - int idx; - switch (type) { - case AUTO_PIN_HP_OUT: - name = "Headphone"; - idx = i; - break; - case AUTO_PIN_SPEAKER_OUT: - if (num_outs <= 2) { - name = i ? "Bass Speaker" : "Speaker"; - idx = 0; - break; - } - /* Fall through in case of multi speaker outs */ - default: - name = chname[i]; - idx = 0; - break; - } - err = create_controls_idx(codec, name, idx, nid, 3); - if (err < 0) - return err; - } - } - return 0; -} - -static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, - unsigned int dir_mask, unsigned int data); - -/* hook for controlling mic-mute LED GPIO */ -static int stac92xx_capture_sw_put_led(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - int err; - bool mute; - - err = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - if (err <= 0) - return err; - mute = !(ucontrol->value.integer.value[0] && - ucontrol->value.integer.value[1]); - if (spec->mic_mute_led_on != mute) { - spec->mic_mute_led_on = mute; - if (mute) - spec->gpio_data |= spec->mic_mute_led_gpio; - else - spec->gpio_data &= ~spec->mic_mute_led_gpio; - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - } - return err; -} - -static int stac92xx_add_capvol_ctls(struct hda_codec *codec, unsigned long vol, - unsigned long sw, int idx) -{ - struct sigmatel_spec *spec = codec->spec; - struct snd_kcontrol_new *knew; - int err; - - err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx, - "Capture Volume", vol); - if (err < 0) - return err; - - knew = add_control_temp(spec, - &stac92xx_control_templates[STAC_CTL_WIDGET_MUTE], - idx, "Capture Switch", sw); - if (!knew) - return -ENOMEM; - /* add a LED hook for some HP laptops */ - if (spec->mic_mute_led_gpio) - knew->put = stac92xx_capture_sw_put_led; - - return 0; -} - -/* add playback controls from the parsed DAC table */ -static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid; - int err; - int idx; - - err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, - spec->multiout.dac_nids, - cfg->line_out_type); - if (err < 0) - return err; - - if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_HP_SWITCH, - "Headphone as Line Out Switch", - cfg->hp_pins[cfg->hp_outs - 1]); - if (err < 0) - return err; - } - - for (idx = 0; idx < cfg->num_inputs; idx++) { - if (cfg->inputs[idx].type > AUTO_PIN_LINE_IN) - break; - nid = cfg->inputs[idx].pin; - err = stac92xx_add_jack_mode_control(codec, nid, idx); - if (err < 0) - return err; - } - - return 0; -} - -/* add playback controls for Speaker and HP outputs */ -static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, - struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int err; - - err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins, - spec->hp_dacs, AUTO_PIN_HP_OUT); - if (err < 0) - return err; - - err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, - spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT); - if (err < 0) - return err; - - return 0; -} - -/* labels for mono mux outputs */ -static const char * const stac92xx_mono_labels[4] = { - "DAC0", "DAC1", "Mixer", "DAC2" -}; - -/* create mono mux for mono out on capable codecs */ -static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *mono_mux = &spec->private_mono_mux; - int i, num_cons; - hda_nid_t con_lst[ARRAY_SIZE(stac92xx_mono_labels)]; - - num_cons = snd_hda_get_connections(codec, - spec->mono_nid, - con_lst, - HDA_MAX_NUM_INPUTS); - if (num_cons <= 0 || num_cons > ARRAY_SIZE(stac92xx_mono_labels)) - return -EINVAL; - - for (i = 0; i < num_cons; i++) - snd_hda_add_imux_item(mono_mux, stac92xx_mono_labels[i], i, - NULL); - - return stac92xx_add_control(spec, STAC_CTL_WIDGET_MONO_MUX, - "Mono Mux", spec->mono_nid); -} - -/* create PC beep volume controls */ -static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, - hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); - int err, type = STAC_CTL_WIDGET_MUTE_BEEP; - - if (spec->anabeep_nid == nid) - type = STAC_CTL_WIDGET_MUTE; - - /* check for mute support for the the amp */ - if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { - err = stac92xx_add_control(spec, type, - "Beep Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); - if (err < 0) - return err; - } - - /* check to see if there is volume support for the amp */ - if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "Beep Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); - if (err < 0) - return err; - } - return 0; -} - -#ifdef CONFIG_SND_HDA_INPUT_BEEP -#define stac92xx_dig_beep_switch_info snd_ctl_boolean_mono_info - -static int stac92xx_dig_beep_switch_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = codec->beep->enabled; - return 0; -} - -static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]); -} - -static const struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = stac92xx_dig_beep_switch_info, - .get = stac92xx_dig_beep_switch_get, - .put = stac92xx_dig_beep_switch_put, -}; - -static int stac92xx_beep_switch_ctl(struct hda_codec *codec) -{ - return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl, - 0, "Beep Playback Switch", 0); -} -#endif - -static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i, j, err = 0; - - for (i = 0; i < spec->num_muxes; i++) { - hda_nid_t nid; - unsigned int wcaps; - unsigned long val; - - nid = spec->mux_nids[i]; - wcaps = get_wcaps(codec, nid); - if (!(wcaps & AC_WCAP_OUT_AMP)) - continue; - - /* check whether already the same control was created as - * normal Capture Volume. - */ - val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); - for (j = 0; j < spec->num_caps; j++) { - if (spec->capvols[j] == val) - break; - } - if (j < spec->num_caps) - continue; - - err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, i, - "Mux Capture Volume", val); - if (err < 0) - return err; - } - return 0; -}; - -static const char * const stac92xx_spdif_labels[3] = { - "Digital Playback", "Analog Mux 1", "Analog Mux 2", -}; - -static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *spdif_mux = &spec->private_smux; - const char * const *labels = spec->spdif_labels; - int i, num_cons; - hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - - num_cons = snd_hda_get_connections(codec, - spec->smux_nids[0], - con_lst, - HDA_MAX_NUM_INPUTS); - if (num_cons <= 0) - return -EINVAL; - - if (!labels) - labels = stac92xx_spdif_labels; - - for (i = 0; i < num_cons; i++) - snd_hda_add_imux_item(spdif_mux, labels[i], i, NULL); - - return 0; -} - -/* labels for dmic mux inputs */ -static const char * const stac92xx_dmic_labels[5] = { - "Analog Inputs", "Digital Mic 1", "Digital Mic 2", - "Digital Mic 3", "Digital Mic 4" -}; - -static hda_nid_t get_connected_node(struct hda_codec *codec, hda_nid_t mux, - int idx) -{ - hda_nid_t conn[HDA_MAX_NUM_INPUTS]; - int nums; - nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); - if (idx >= 0 && idx < nums) - return conn[idx]; - return 0; -} - -/* look for NID recursively */ -#define get_connection_index(codec, mux, nid) \ - snd_hda_get_conn_index(codec, mux, nid, 1) - -/* create a volume assigned to the given pin (only if supported) */ -/* return 1 if the volume control is created */ -static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid, - const char *label, int idx, int direction) -{ - unsigned int caps, nums; - char name[32]; - int err; - - if (direction == HDA_OUTPUT) - caps = AC_WCAP_OUT_AMP; - else - caps = AC_WCAP_IN_AMP; - if (!(get_wcaps(codec, nid) & caps)) - return 0; - caps = query_amp_caps(codec, nid, direction); - nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; - if (!nums) - return 0; - snprintf(name, sizeof(name), "%s Capture Volume", label); - err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, direction)); - if (err < 0) - return err; - return 1; -} - -/* create playback/capture controls for input pins on dmic capable codecs */ -static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *imux = &spec->private_imux; - struct hda_input_mux *dimux = &spec->private_dimux; - int err, i; - unsigned int def_conf; - - snd_hda_add_imux_item(dimux, stac92xx_dmic_labels[0], 0, NULL); - - for (i = 0; i < spec->num_dmics; i++) { - hda_nid_t nid; - int index, type_idx; - char label[32]; - - nid = spec->dmic_nids[i]; - if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) - continue; - def_conf = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) - continue; - - index = get_connection_index(codec, spec->dmux_nids[0], nid); - if (index < 0) - continue; - - snd_hda_get_pin_label(codec, nid, &spec->autocfg, - label, sizeof(label), NULL); - snd_hda_add_imux_item(dimux, label, index, &type_idx); - if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) - snd_hda_add_imux_item(imux, label, index, &type_idx); - - err = create_elem_capture_vol(codec, nid, label, type_idx, - HDA_INPUT); - if (err < 0) - return err; - if (!err) { - err = create_elem_capture_vol(codec, nid, label, - type_idx, HDA_OUTPUT); - if (err < 0) - return err; - if (!err) { - nid = get_connected_node(codec, - spec->dmux_nids[0], index); - if (nid) - err = create_elem_capture_vol(codec, - nid, label, - type_idx, HDA_INPUT); - if (err < 0) - return err; - } - } - } - - return 0; -} - -static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *fixed, hda_nid_t *ext, hda_nid_t *dock) -{ - unsigned int cfg; - unsigned int type; - - if (!nid) - return 0; - cfg = snd_hda_codec_get_pincfg(codec, nid); - type = get_defcfg_device(cfg); - switch (snd_hda_get_input_pin_attr(cfg)) { - case INPUT_PIN_ATTR_INT: - if (*fixed) - return 1; /* already occupied */ - if (type != AC_JACK_MIC_IN) - return 1; /* invalid type */ - *fixed = nid; - break; - case INPUT_PIN_ATTR_UNUSED: - break; - case INPUT_PIN_ATTR_DOCK: - if (*dock) - return 1; /* already occupied */ - if (type != AC_JACK_MIC_IN && type != AC_JACK_LINE_IN) - return 1; /* invalid type */ - *dock = nid; - break; - default: - if (*ext) - return 1; /* already occupied */ - if (type != AC_JACK_MIC_IN) - return 1; /* invalid type */ - *ext = nid; - break; - } - return 0; -} - -static int set_mic_route(struct hda_codec *codec, - struct sigmatel_mic_route *mic, - hda_nid_t pin) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; - - mic->pin = pin; - if (pin == 0) - return 0; - for (i = 0; i < cfg->num_inputs; i++) { - if (pin == cfg->inputs[i].pin) - break; - } - if (i < cfg->num_inputs && cfg->inputs[i].type == AUTO_PIN_MIC) { - /* analog pin */ - i = get_connection_index(codec, spec->mux_nids[0], pin); - if (i < 0) - return -1; - mic->mux_idx = i; - mic->dmux_idx = -1; - if (spec->dmux_nids) - mic->dmux_idx = get_connection_index(codec, - spec->dmux_nids[0], - spec->mux_nids[0]); - } else if (spec->dmux_nids) { - /* digital pin */ - i = get_connection_index(codec, spec->dmux_nids[0], pin); - if (i < 0) - return -1; - mic->dmux_idx = i; - mic->mux_idx = -1; - if (spec->mux_nids) - mic->mux_idx = get_connection_index(codec, - spec->mux_nids[0], - spec->dmux_nids[0]); - } - return 0; -} - -/* return non-zero if the device is for automatic mic switch */ -static int stac_check_auto_mic(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t fixed, ext, dock; - int i; - - fixed = ext = dock = 0; - for (i = 0; i < cfg->num_inputs; i++) - if (check_mic_pin(codec, cfg->inputs[i].pin, - &fixed, &ext, &dock)) - return 0; - for (i = 0; i < spec->num_dmics; i++) - if (check_mic_pin(codec, spec->dmic_nids[i], - &fixed, &ext, &dock)) - return 0; - if (!fixed || (!ext && !dock)) - return 0; /* no input to switch */ - if (!is_jack_detectable(codec, ext)) - return 0; /* no unsol support */ - if (set_mic_route(codec, &spec->ext_mic, ext) || - set_mic_route(codec, &spec->int_mic, fixed) || - set_mic_route(codec, &spec->dock_mic, dock)) - return 0; /* something is wrong */ - return 1; -} - -/* create playback/capture controls for input pins */ -static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *imux = &spec->private_imux; - int i, j; - const char *label; - - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; - int index, err, type_idx; - - index = -1; - for (j = 0; j < spec->num_muxes; j++) { - index = get_connection_index(codec, spec->mux_nids[j], - nid); - if (index >= 0) - break; - } - if (index < 0) - continue; - - label = hda_get_autocfg_input_label(codec, cfg, i); - snd_hda_add_imux_item(imux, label, index, &type_idx); - - err = create_elem_capture_vol(codec, nid, - label, type_idx, - HDA_INPUT); - if (err < 0) - return err; - } - spec->num_analog_muxes = imux->num_items; - - if (imux->num_items) { - /* - * Set the current input for the muxes. - * The STAC9221 has two input muxes with identical source - * NID lists. Hopefully this won't get confused. - */ - for (i = 0; i < spec->num_muxes; i++) { - snd_hda_codec_write_cache(codec, spec->mux_nids[i], 0, - AC_VERB_SET_CONNECT_SEL, - imux->items[0].index); - } - } - - return 0; -} - -static void stac92xx_auto_init_multi_out(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->autocfg.line_outs; i++) { - hda_nid_t nid = spec->autocfg.line_out_pins[i]; - stac92xx_auto_set_pinctl(codec, nid, AC_PINCTL_OUT_EN); - } -} - -static void stac92xx_auto_init_hp_out(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->autocfg.hp_outs; i++) { - hda_nid_t pin; - pin = spec->autocfg.hp_pins[i]; - if (pin) /* connect to front */ - stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); - } - for (i = 0; i < spec->autocfg.speaker_outs; i++) { - hda_nid_t pin; - pin = spec->autocfg.speaker_pins[i]; - if (pin) /* connect to front */ - stac92xx_auto_set_pinctl(codec, pin, AC_PINCTL_OUT_EN); - } -} - -static int is_dual_headphones(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int i, valid_hps; - - if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT || - spec->autocfg.hp_outs <= 1) - return 0; - valid_hps = 0; - for (i = 0; i < spec->autocfg.hp_outs; i++) { - hda_nid_t nid = spec->autocfg.hp_pins[i]; - unsigned int cfg = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_location(cfg) & AC_JACK_LOC_SEPARATE) - continue; - valid_hps++; - } - return (valid_hps > 1); -} - - -static int stac92xx_parse_auto_config(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t dig_out = 0, dig_in = 0; - int hp_swap = 0; - int i, err; - - if ((err = snd_hda_parse_pin_def_config(codec, - &spec->autocfg, - spec->dmic_nids)) < 0) - return err; - if (! spec->autocfg.line_outs) - return -EINVAL; /* can't find valid pin config */ - - /* If we have no real line-out pin and multiple hp-outs, HPs should - * be set up as multi-channel outputs. - */ - if (is_dual_headphones(codec)) { - /* Copy hp_outs to line_outs, backup line_outs in - * speaker_outs so that the following routines can handle - * HP pins as primary outputs. - */ - snd_printdd("stac92xx: Enabling multi-HPs workaround\n"); - memcpy(spec->autocfg.speaker_pins, spec->autocfg.line_out_pins, - sizeof(spec->autocfg.line_out_pins)); - spec->autocfg.speaker_outs = spec->autocfg.line_outs; - memcpy(spec->autocfg.line_out_pins, spec->autocfg.hp_pins, - sizeof(spec->autocfg.hp_pins)); - spec->autocfg.line_outs = spec->autocfg.hp_outs; - spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; - spec->autocfg.hp_outs = 0; - hp_swap = 1; - } - if (spec->autocfg.mono_out_pin) { - int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & - (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); - u32 caps = query_amp_caps(codec, - spec->autocfg.mono_out_pin, dir); - hda_nid_t conn_list[1]; - - /* get the mixer node and then the mono mux if it exists */ - if (snd_hda_get_connections(codec, - spec->autocfg.mono_out_pin, conn_list, 1) && - snd_hda_get_connections(codec, conn_list[0], - conn_list, 1) > 0) { - - int wcaps = get_wcaps(codec, conn_list[0]); - int wid_type = get_wcaps_type(wcaps); - /* LR swap check, some stac925x have a mux that - * changes the DACs output path instead of the - * mono-mux path. - */ - if (wid_type == AC_WID_AUD_SEL && - !(wcaps & AC_WCAP_LR_SWAP)) - spec->mono_nid = conn_list[0]; - } - if (dir) { - hda_nid_t nid = spec->autocfg.mono_out_pin; - - /* most mono outs have a least a mute/unmute switch */ - dir = (dir & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "Mono Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); - if (err < 0) - return err; - /* check for volume support for the amp */ - if ((caps & AC_AMPCAP_NUM_STEPS) - >> AC_AMPCAP_NUM_STEPS_SHIFT) { - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_VOL, - "Mono Playback Volume", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); - if (err < 0) - return err; - } - } - - stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin, - AC_PINCTL_OUT_EN); - } - - if (!spec->multiout.num_dacs) { - err = stac92xx_auto_fill_dac_nids(codec); - if (err < 0) - return err; - err = stac92xx_auto_create_multi_out_ctls(codec, - &spec->autocfg); - if (err < 0) - return err; - } - - /* setup analog beep controls */ - if (spec->anabeep_nid > 0) { - err = stac92xx_auto_create_beep_ctls(codec, - spec->anabeep_nid); - if (err < 0) - return err; - } - - /* setup digital beep controls and input device */ -#ifdef CONFIG_SND_HDA_INPUT_BEEP - if (spec->digbeep_nid > 0) { - hda_nid_t nid = spec->digbeep_nid; - unsigned int caps; - - err = stac92xx_auto_create_beep_ctls(codec, nid); - if (err < 0) - return err; - err = snd_hda_attach_beep_device(codec, nid); - if (err < 0) - return err; - if (codec->beep) { - /* IDT/STAC codecs have linear beep tone parameter */ - codec->beep->linear_tone = spec->linear_tone_beep; - /* if no beep switch is available, make its own one */ - caps = query_amp_caps(codec, nid, HDA_OUTPUT); - if (!(caps & AC_AMPCAP_MUTE)) { - err = stac92xx_beep_switch_ctl(codec); - if (err < 0) - return err; - } - } - } -#endif - - err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); - if (err < 0) - return err; - - /* All output parsing done, now restore the swapped hp pins */ - if (hp_swap) { - memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, - sizeof(spec->autocfg.hp_pins)); - spec->autocfg.hp_outs = spec->autocfg.line_outs; - spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; - spec->autocfg.line_outs = 0; - } - - if (stac_check_auto_mic(codec)) { - spec->auto_mic = 1; - /* only one capture for auto-mic */ - spec->num_adcs = 1; - spec->num_caps = 1; - spec->num_muxes = 1; - } - - for (i = 0; i < spec->num_caps; i++) { - err = stac92xx_add_capvol_ctls(codec, spec->capvols[i], - spec->capsws[i], i); - if (err < 0) - return err; - } - - err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); - if (err < 0) - return err; - - if (spec->mono_nid > 0) { - err = stac92xx_auto_create_mono_output_ctls(codec); - if (err < 0) - return err; - } - if (spec->num_dmics > 0 && !spec->dinput_mux) - if ((err = stac92xx_auto_create_dmic_input_ctls(codec, - &spec->autocfg)) < 0) - return err; - if (spec->num_muxes > 0) { - err = stac92xx_auto_create_mux_input_ctls(codec); - if (err < 0) - return err; - } - if (spec->num_smuxes > 0) { - err = stac92xx_auto_create_spdif_mux_ctls(codec); - if (err < 0) - return err; - } - - err = stac92xx_add_input_source(spec); - if (err < 0) - return err; - - spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->multiout.max_channels > 2) - spec->surr_switch = 1; - - /* find digital out and in converters */ - for (i = codec->start_nid; i < codec->start_nid + codec->num_nodes; i++) { - unsigned int wid_caps = get_wcaps(codec, i); - if (wid_caps & AC_WCAP_DIGITAL) { - switch (get_wcaps_type(wid_caps)) { - case AC_WID_AUD_OUT: - if (!dig_out) - dig_out = i; - break; - case AC_WID_AUD_IN: - if (!dig_in) - dig_in = i; - break; - } - } - } - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = dig_out; - if (dig_in && spec->autocfg.dig_in_pin) - spec->dig_in_nid = dig_in; - - if (spec->kctls.list) - spec->mixers[spec->num_mixers++] = spec->kctls.list; - - spec->input_mux = &spec->private_imux; - if (!spec->dinput_mux) - spec->dinput_mux = &spec->private_dimux; - spec->sinput_mux = &spec->private_smux; - spec->mono_mux = &spec->private_mono_mux; - return 0; -} - -/* add playback controls for HP output */ -static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, - struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - hda_nid_t pin = cfg->hp_pins[0]; - - if (! pin) - return 0; - - if (is_jack_detectable(codec, pin)) - spec->hp_detect = 1; - - return 0; -} - -/* add playback controls for LFE output */ -static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, - struct auto_pin_cfg *cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int err; - hda_nid_t lfe_pin = 0x0; - int i; - - /* - * search speaker outs and line outs for a mono speaker pin - * with an amp. If one is found, add LFE controls - * for it. - */ - for (i = 0; i < spec->autocfg.speaker_outs && lfe_pin == 0x0; i++) { - hda_nid_t pin = spec->autocfg.speaker_pins[i]; - unsigned int wcaps = get_wcaps(codec, pin); - wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); - if (wcaps == AC_WCAP_OUT_AMP) - /* found a mono speaker with an amp, must be lfe */ - lfe_pin = pin; - } - - /* if speaker_outs is 0, then speakers may be in line_outs */ - if (lfe_pin == 0 && spec->autocfg.speaker_outs == 0) { - for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { - hda_nid_t pin = spec->autocfg.line_out_pins[i]; - unsigned int defcfg; - defcfg = snd_hda_codec_get_pincfg(codec, pin); - if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { - unsigned int wcaps = get_wcaps(codec, pin); - wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); - if (wcaps == AC_WCAP_OUT_AMP) - /* found a mono speaker with an amp, - must be lfe */ - lfe_pin = pin; - } - } - } - - if (lfe_pin) { - err = create_controls(codec, "LFE", lfe_pin, 1); - if (err < 0) - return err; - } - - return 0; -} - -static int stac9200_parse_auto_config(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int err; - - if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0) - return err; - - if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) - return err; - - if ((err = stac9200_auto_create_hp_ctls(codec, &spec->autocfg)) < 0) - return err; - - if ((err = stac9200_auto_create_lfe_ctls(codec, &spec->autocfg)) < 0) - return err; - - if (spec->num_muxes > 0) { - err = stac92xx_auto_create_mux_input_ctls(codec); - if (err < 0) - return err; - } - - err = stac92xx_add_input_source(spec); - if (err < 0) - return err; - - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = 0x05; - if (spec->autocfg.dig_in_pin) - spec->dig_in_nid = 0x04; - - if (spec->kctls.list) - spec->mixers[spec->num_mixers++] = spec->kctls.list; - - spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; - - return 0; -} - -/* - * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a - * funky external mute control using GPIO pins. - */ - -static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, - unsigned int dir_mask, unsigned int data) -{ - unsigned int gpiostate, gpiomask, gpiodir; - - snd_printdd("%s msk %x dir %x gpio %x\n", __func__, mask, dir_mask, data); - - gpiostate = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0); - gpiostate = (gpiostate & ~dir_mask) | (data & dir_mask); - - gpiomask = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_MASK, 0); - gpiomask |= mask; - - gpiodir = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DIRECTION, 0); - gpiodir |= dir_mask; - - /* Configure GPIOx as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); - - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, gpiomask); - snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, gpiodir); /* sync */ - - msleep(1); - - snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ -} - -static int stac_add_event(struct hda_codec *codec, hda_nid_t nid, - unsigned char type, int data) -{ - struct hda_jack_tbl *event; - - event = snd_hda_jack_tbl_new(codec, nid); - if (!event) - return -ENOMEM; - event->action = type; - event->private_data = data; - - return 0; -} - -static void handle_unsol_event(struct hda_codec *codec, - struct hda_jack_tbl *event); - -/* check if given nid is a valid pin and no other events are assigned - * to it. If OK, assign the event, set the unsol flag, and returns 1. - * Otherwise, returns zero. - */ -static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, - unsigned int type) -{ - struct hda_jack_tbl *event; - - if (!is_jack_detectable(codec, nid)) - return 0; - event = snd_hda_jack_tbl_new(codec, nid); - if (!event) - return -ENOMEM; - if (event->action && event->action != type) - return 0; - event->action = type; - event->callback = handle_unsol_event; - snd_hda_jack_detect_enable(codec, nid, 0); - return 1; -} - -static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) -{ - int i; - for (i = 0; i < cfg->hp_outs; i++) - if (cfg->hp_pins[i] == nid) - return 1; /* nid is a HP-Out */ - for (i = 0; i < cfg->line_outs; i++) - if (cfg->line_out_pins[i] == nid) - return 1; /* nid is a line-Out */ - return 0; /* nid is not a HP-Out */ -}; - -static void stac92xx_power_down(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - /* power down inactive DACs */ - const hda_nid_t *dac; - for (dac = spec->dac_list; *dac; dac++) - if (!check_all_dac_nids(spec, *dac)) - snd_hda_codec_write(codec, *dac, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); -} - -static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, - int enable); - -static inline bool get_int_hint(struct hda_codec *codec, const char *key, - int *valp) -{ - return !snd_hda_get_int_hint(codec, key, valp); -} - -/* override some hints from the hwdep entry */ -static void stac_store_hints(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - int val; - - val = snd_hda_get_bool_hint(codec, "hp_detect"); - if (val >= 0) - spec->hp_detect = val; - if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) { - spec->eapd_mask = spec->gpio_dir = spec->gpio_data = - spec->gpio_mask; - } - if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) - spec->gpio_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) - spec->gpio_dir &= spec->gpio_mask; - if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) - spec->eapd_mask &= spec->gpio_mask; - if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) - spec->gpio_mute &= spec->gpio_mask; - val = snd_hda_get_bool_hint(codec, "eapd_switch"); - if (val >= 0) - spec->eapd_switch = val; -} - -static void stac_issue_unsol_events(struct hda_codec *codec, int num_pins, - const hda_nid_t *pins) -{ - while (num_pins--) - stac_issue_unsol_event(codec, *pins++); -} - -/* fake event to set up pins */ -static void stac_fake_hp_events(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (spec->autocfg.hp_outs) - stac_issue_unsol_events(codec, spec->autocfg.hp_outs, - spec->autocfg.hp_pins); - if (spec->autocfg.line_outs && - spec->autocfg.line_out_pins[0] != spec->autocfg.hp_pins[0]) - stac_issue_unsol_events(codec, spec->autocfg.line_outs, - spec->autocfg.line_out_pins); -} - -static int stac92xx_init(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int gpio; - int i; - - if (spec->init) - snd_hda_sequence_write(codec, spec->init); - - snd_hda_apply_verbs(codec); - - /* power down adcs initially */ - if (spec->powerdown_adcs) - for (i = 0; i < spec->num_adcs; i++) - snd_hda_codec_write(codec, - spec->adc_nids[i], 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* override some hints */ - stac_store_hints(codec); - - /* set up GPIO */ - gpio = spec->gpio_data; - /* turn on EAPD statically when spec->eapd_switch isn't set. - * otherwise, unsol event will turn it on/off dynamically - */ - if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); - - /* set up pins */ - if (spec->hp_detect) { - /* Enable unsolicited responses on the HP widget */ - for (i = 0; i < cfg->hp_outs; i++) { - hda_nid_t nid = cfg->hp_pins[i]; - enable_pin_detect(codec, nid, STAC_HP_EVENT); - } - if (cfg->line_out_type == AUTO_PIN_LINE_OUT && - cfg->speaker_outs > 0) { - /* enable pin-detect for line-outs as well */ - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t nid = cfg->line_out_pins[i]; - enable_pin_detect(codec, nid, STAC_LO_EVENT); - } - } - - /* force to enable the first line-out; the others are set up - * in unsol_event - */ - stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], - AC_PINCTL_OUT_EN); - /* fake event to set up pins */ - stac_fake_hp_events(codec); - } else { - stac92xx_auto_init_multi_out(codec); - stac92xx_auto_init_hp_out(codec); - for (i = 0; i < cfg->hp_outs; i++) - stac_toggle_power_map(codec, cfg->hp_pins[i], 1); - } - if (spec->auto_mic) { - /* initialize connection to analog input */ - if (spec->dmux_nids) - snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, 0); - if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT)) - stac_issue_unsol_event(codec, spec->ext_mic.pin); - if (enable_pin_detect(codec, spec->dock_mic.pin, - STAC_MIC_EVENT)) - stac_issue_unsol_event(codec, spec->dock_mic.pin); - } - for (i = 0; i < cfg->num_inputs; i++) { - hda_nid_t nid = cfg->inputs[i].pin; - int type = cfg->inputs[i].type; - unsigned int pinctl, conf; - if (type == AUTO_PIN_MIC) { - /* for mic pins, force to initialize */ - pinctl = snd_hda_get_default_vref(codec, nid); - pinctl |= AC_PINCTL_IN_EN; - stac92xx_auto_set_pinctl(codec, nid, pinctl); - } else { - pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - /* if PINCTL already set then skip */ - /* Also, if both INPUT and OUTPUT are set, - * it must be a BIOS bug; need to override, too - */ - if (!(pinctl & AC_PINCTL_IN_EN) || - (pinctl & AC_PINCTL_OUT_EN)) { - pinctl &= ~AC_PINCTL_OUT_EN; - pinctl |= AC_PINCTL_IN_EN; - stac92xx_auto_set_pinctl(codec, nid, pinctl); - } - } - conf = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { - if (enable_pin_detect(codec, nid, STAC_INSERT_EVENT)) - stac_issue_unsol_event(codec, nid); - } - } - for (i = 0; i < spec->num_dmics; i++) - stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], - AC_PINCTL_IN_EN); - if (cfg->dig_out_pins[0]) - stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0], - AC_PINCTL_OUT_EN); - if (cfg->dig_in_pin) - stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, - AC_PINCTL_IN_EN); - for (i = 0; i < spec->num_pwrs; i++) { - hda_nid_t nid = spec->pwr_nids[i]; - unsigned int pinctl, def_conf; - - def_conf = snd_hda_codec_get_pincfg(codec, nid); - def_conf = get_defcfg_connect(def_conf); - if (def_conf == AC_JACK_PORT_NONE) { - /* power off unused ports */ - stac_toggle_power_map(codec, nid, 0); - continue; - } - if (def_conf == AC_JACK_PORT_FIXED) { - /* no need for jack detection for fixed pins */ - stac_toggle_power_map(codec, nid, 1); - continue; - } - /* power on when no jack detection is available */ - /* or when the VREF is used for controlling LED */ - if (!spec->hp_detect || - spec->vref_mute_led_nid == nid || - !is_jack_detectable(codec, nid)) { - stac_toggle_power_map(codec, nid, 1); - continue; - } - - if (is_nid_out_jack_pin(cfg, nid)) - continue; /* already has an unsol event */ - - pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - /* outputs are only ports capable of power management - * any attempts on powering down a input port cause the - * referenced VREF to act quirky. - */ - if (pinctl & AC_PINCTL_IN_EN) { - stac_toggle_power_map(codec, nid, 1); - continue; - } - if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) { - stac_issue_unsol_event(codec, nid); - continue; - } - /* none of the above, turn the port OFF */ - stac_toggle_power_map(codec, nid, 0); - } - - /* sync mute LED */ - if (spec->gpio_led) { - if (spec->vmaster_mute.hook) - snd_hda_sync_vmaster_hook(&spec->vmaster_mute); - else /* the very first init call doesn't have vmaster yet */ - stac92xx_update_led_status(codec, false); - } - - /* sync the power-map */ - if (spec->num_pwrs) - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_IDT_SET_POWER_MAP, - spec->power_map_bits); - if (spec->dac_list) - stac92xx_power_down(codec); - return 0; -} - -static void stac92xx_free_kctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (spec->kctls.list) { - struct snd_kcontrol_new *kctl = spec->kctls.list; - int i; - for (i = 0; i < spec->kctls.used; i++) - kfree(kctl[i].name); - } - snd_array_free(&spec->kctls); -} - -static void stac92xx_shutup_pins(struct hda_codec *codec) -{ - unsigned int i, def_conf; - - if (codec->bus->shutdown) - return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); - def_conf = snd_hda_codec_get_pincfg(codec, pin->nid); - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) - snd_hda_set_pin_ctl(codec, pin->nid, 0); - } -} - -static void stac92xx_shutup(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - stac92xx_shutup_pins(codec); - - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); -} - -static void stac92xx_free(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (! spec) - return; - - kfree(spec); - snd_hda_detach_beep_device(codec); -} - -static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, - unsigned int flag) -{ - unsigned int old_ctl, pin_ctl; - - pin_ctl = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - - if (pin_ctl & AC_PINCTL_IN_EN) { - /* - * we need to check the current set-up direction of - * shared input pins since they can be switched via - * "xxx as Output" mixer switch - */ - struct sigmatel_spec *spec = codec->spec; - if (nid == spec->line_switch || nid == spec->mic_switch) - return; - } - - old_ctl = pin_ctl; - /* if setting pin direction bits, clear the current - direction bits first */ - if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)) - pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - - pin_ctl |= flag; - if (old_ctl != pin_ctl) - snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl); -} - -static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, - unsigned int flag) -{ - unsigned int pin_ctl = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - if (pin_ctl & flag) - snd_hda_set_pin_ctl_cache(codec, nid, pin_ctl & ~flag); -} - -static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) -{ - if (!nid) - return 0; - return snd_hda_jack_detect(codec, nid); -} - -static void stac92xx_line_out_detect(struct hda_codec *codec, - int presence) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; - - if (cfg->speaker_outs == 0) - return; - - for (i = 0; i < cfg->line_outs; i++) { - if (presence) - break; - presence = get_pin_presence(codec, cfg->line_out_pins[i]); - if (presence) { - unsigned int pinctl; - pinctl = snd_hda_codec_read(codec, - cfg->line_out_pins[i], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (pinctl & AC_PINCTL_IN_EN) - presence = 0; /* mic- or line-input */ - } - } - - if (presence) { - /* disable speakers */ - for (i = 0; i < cfg->speaker_outs; i++) - stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], - AC_PINCTL_OUT_EN); - if (spec->eapd_mask && spec->eapd_switch) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); - } else { - /* enable speakers */ - for (i = 0; i < cfg->speaker_outs; i++) - stac92xx_set_pinctl(codec, cfg->speaker_pins[i], - AC_PINCTL_OUT_EN); - if (spec->eapd_mask && spec->eapd_switch) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data | - spec->eapd_mask); - } -} - -/* return non-zero if the hp-pin of the given array index isn't - * a jack-detection target - */ -static int no_hp_sensing(struct sigmatel_spec *spec, int i) -{ - struct auto_pin_cfg *cfg = &spec->autocfg; - - /* ignore sensing of shared line and mic jacks */ - if (cfg->hp_pins[i] == spec->line_switch) - return 1; - if (cfg->hp_pins[i] == spec->mic_switch) - return 1; - /* ignore if the pin is set as line-out */ - if (cfg->hp_pins[i] == spec->hp_switch) - return 1; - return 0; -} - -static void stac92xx_hp_detect(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - int i, presence; - - presence = 0; - if (spec->gpio_mute) - presence = !(snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0) & spec->gpio_mute); - - for (i = 0; i < cfg->hp_outs; i++) { - if (presence) - break; - if (no_hp_sensing(spec, i)) - continue; - presence = get_pin_presence(codec, cfg->hp_pins[i]); - if (presence) { - unsigned int pinctl; - pinctl = snd_hda_codec_read(codec, cfg->hp_pins[i], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (pinctl & AC_PINCTL_IN_EN) - presence = 0; /* mic- or line-input */ - } - } - - if (presence) { - /* disable lineouts */ - if (spec->hp_switch) - stac92xx_reset_pinctl(codec, spec->hp_switch, - AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->line_outs; i++) - stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], - AC_PINCTL_OUT_EN); - } else { - /* enable lineouts */ - if (spec->hp_switch) - stac92xx_set_pinctl(codec, spec->hp_switch, - AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->line_outs; i++) - stac92xx_set_pinctl(codec, cfg->line_out_pins[i], - AC_PINCTL_OUT_EN); - } - stac92xx_line_out_detect(codec, presence); - /* toggle hp outs */ - for (i = 0; i < cfg->hp_outs; i++) { - unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; - if (no_hp_sensing(spec, i)) - continue; - if (1 /*presence*/) - stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); -#if 0 /* FIXME */ -/* Resetting the pinctl like below may lead to (a sort of) regressions - * on some devices since they use the HP pin actually for line/speaker - * outs although the default pin config shows a different pin (that is - * wrong and useless). - * - * So, it's basically a problem of default pin configs, likely a BIOS issue. - * But, disabling the code below just works around it, and I'm too tired of - * bug reports with such devices... - */ - else - stac92xx_reset_pinctl(codec, cfg->hp_pins[i], val); -#endif /* FIXME */ - } -} - -static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, - int enable) -{ - struct sigmatel_spec *spec = codec->spec; - unsigned int idx, val; - - for (idx = 0; idx < spec->num_pwrs; idx++) { - if (spec->pwr_nids[idx] == nid) - break; - } - if (idx >= spec->num_pwrs) - return; - - idx = 1 << idx; - - val = spec->power_map_bits; - if (enable) - val &= ~idx; - else - val |= idx; - - /* power down unused output ports */ - if (val != spec->power_map_bits) { - spec->power_map_bits = val; - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_IDT_SET_POWER_MAP, val); - } -} - -static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid) -{ - stac_toggle_power_map(codec, nid, get_pin_presence(codec, nid)); -} + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021c, + "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021f, + "Dell Inspiron", STAC_9205_DELL_M44), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, + "Dell Vostro 1500", STAC_9205_DELL_M42), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0229, + "Dell Vostro 1700", STAC_9205_DELL_M42), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0560, "Gateway T6834c", STAC_9205_EAPD), + SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), + {} /* terminator */ +}; -/* get the pin connection (fixed, none, etc) */ -static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) +static int stac_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int cfg; + int err; - cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); - return get_defcfg_connect(cfg); -} + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + if (err < 0) + return err; -static int stac92xx_connected_ports(struct hda_codec *codec, - const hda_nid_t *nids, int num_nids) -{ - struct sigmatel_spec *spec = codec->spec; - int idx, num; - unsigned int def_conf; - - for (num = 0; num < num_nids; num++) { - for (idx = 0; idx < spec->num_pins; idx++) - if (spec->pin_nids[idx] == nids[num]) - break; - if (idx >= spec->num_pins) - break; - def_conf = stac_get_defcfg_connect(codec, idx); - if (def_conf == AC_JACK_PORT_NONE) - break; - } - return num; -} + /* add hooks */ + spec->gen.pcm_playback_hook = stac_playback_pcm_hook; + spec->gen.pcm_capture_hook = stac_capture_pcm_hook; -static void stac92xx_mic_detect(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct sigmatel_mic_route *mic; + spec->gen.automute_hook = stac_update_outputs; + spec->gen.hp_automute_hook = stac_hp_automute; + spec->gen.line_automute_hook = stac_line_automute; - if (get_pin_presence(codec, spec->ext_mic.pin)) - mic = &spec->ext_mic; - else if (get_pin_presence(codec, spec->dock_mic.pin)) - mic = &spec->dock_mic; - else - mic = &spec->int_mic; - if (mic->dmux_idx >= 0) - snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, - mic->dmux_idx); - if (mic->mux_idx >= 0) - snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0, - AC_VERB_SET_CONNECT_SEL, - mic->mux_idx); -} + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); + if (err < 0) + return err; -static void handle_unsol_event(struct hda_codec *codec, - struct hda_jack_tbl *event) -{ - struct sigmatel_spec *spec = codec->spec; - int data; + /* minimum value is actually mute */ + spec->gen.vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; - switch (event->action) { - case STAC_HP_EVENT: - case STAC_LO_EVENT: - stac92xx_hp_detect(codec); - break; - case STAC_MIC_EVENT: - stac92xx_mic_detect(codec); - break; + /* setup analog beep controls */ + if (spec->anabeep_nid > 0) { + err = stac_auto_create_beep_ctls(codec, + spec->anabeep_nid); + if (err < 0) + return err; } - switch (event->action) { - case STAC_HP_EVENT: - case STAC_LO_EVENT: - case STAC_MIC_EVENT: - case STAC_INSERT_EVENT: - case STAC_PWR_EVENT: - if (spec->num_pwrs > 0) - stac92xx_pin_sense(codec, event->nid); - - switch (codec->subsystem_id) { - case 0x103c308f: - if (event->nid == 0xb) { - int pin = AC_PINCTL_IN_EN; - - if (get_pin_presence(codec, 0xa) - && get_pin_presence(codec, 0xb)) - pin |= AC_PINCTL_VREF_80; - if (!get_pin_presence(codec, 0xb)) - pin |= AC_PINCTL_VREF_80; - - /* toggle VREF state based on mic + hp pin - * status - */ - stac92xx_auto_set_pinctl(codec, 0x0a, pin); + /* setup digital beep controls and input device */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (spec->digbeep_nid > 0) { + hda_nid_t nid = spec->digbeep_nid; + unsigned int caps; + + err = stac_auto_create_beep_ctls(codec, nid); + if (err < 0) + return err; + err = snd_hda_attach_beep_device(codec, nid); + if (err < 0) + return err; + if (codec->beep) { + /* IDT/STAC codecs have linear beep tone parameter */ + codec->beep->linear_tone = spec->linear_tone_beep; + /* if no beep switch is available, make its own one */ + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + if (!(caps & AC_AMPCAP_MUTE)) { + err = stac_beep_switch_ctl(codec); + if (err < 0) + return err; } } - break; - case STAC_VREF_EVENT: - data = snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_GET_GPIO_DATA, 0); - /* toggle VREF state based on GPIOx status */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, - !!(data & (1 << event->private_data))); - break; } -} +#endif -static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid) -{ - struct hda_jack_tbl *event = snd_hda_jack_tbl_get(codec, nid); - if (!event) - return; - handle_unsol_event(codec, event); + if (spec->gpio_led) + spec->gen.vmaster_mute.hook = stac_vmaster_hook; + + if (spec->aloopback_ctl && + snd_hda_get_bool_hint(codec, "loopback") == 1) { + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, spec->aloopback_ctl)) + return -ENOMEM; + } + + stac_init_power_map(codec); + + return 0; } -static void set_hp_led_gpio(struct hda_codec *codec) + +static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; unsigned int gpio; + int i; - if (spec->gpio_led) - return; + /* override some hints */ + stac_store_hints(codec); - gpio = snd_hda_param_read(codec, codec->afg, AC_PAR_GPIO_CAP); - gpio &= AC_GPIO_IO_COUNT; - if (gpio > 3) - spec->gpio_led = 0x08; /* GPIO 3 */ - else - spec->gpio_led = 0x01; /* GPIO 0 */ -} + /* set up GPIO */ + gpio = spec->gpio_data; + /* turn on EAPD statically when spec->eapd_switch isn't set. + * otherwise, unsol event will turn it on/off dynamically + */ + if (!spec->eapd_switch) + gpio |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); -/* - * This method searches for the mute LED GPIO configuration - * provided as OEM string in SMBIOS. The format of that string - * is HP_Mute_LED_P_G or HP_Mute_LED_P - * where P can be 0 or 1 and defines mute LED GPIO control state (low/high) - * that corresponds to the NOT muted state of the master volume - * and G is the index of the GPIO to use as the mute LED control (0..9) - * If _G portion is missing it is assigned based on the codec ID - * - * So, HP B-series like systems may have HP_Mute_LED_0 (current models) - * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings - * - * - * The dv-series laptops don't seem to have the HP_Mute_LED* strings in - * SMBIOS - at least the ones I have seen do not have them - which include - * my own system (HP Pavilion dv6-1110ax) and my cousin's - * HP Pavilion dv9500t CTO. - * Need more information on whether it is true across the entire series. - * -- kunal - */ -static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) -{ - struct sigmatel_spec *spec = codec->spec; - const struct dmi_device *dev = NULL; + snd_hda_gen_init(codec); - if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { - get_int_hint(codec, "gpio_led_polarity", - &spec->gpio_led_polarity); - return 1; - } + /* sync the power-map */ + if (spec->num_pwrs) + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_IDT_SET_POWER_MAP, + spec->power_map_bits); - while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { - if (sscanf(dev->name, "HP_Mute_LED_%d_%x", - &spec->gpio_led_polarity, - &spec->gpio_led) == 2) { - unsigned int max_gpio; - max_gpio = snd_hda_param_read(codec, codec->afg, - AC_PAR_GPIO_CAP); - max_gpio &= AC_GPIO_IO_COUNT; - if (spec->gpio_led < max_gpio) - spec->gpio_led = 1 << spec->gpio_led; - else - spec->vref_mute_led_nid = spec->gpio_led; - return 1; - } - if (sscanf(dev->name, "HP_Mute_LED_%d", - &spec->gpio_led_polarity) == 1) { - set_hp_led_gpio(codec); - return 1; - } - /* BIOS bug: unfilled OEM string */ - if (strstr(dev->name, "HP_Mute_LED_P_G")) { - set_hp_led_gpio(codec); - if (default_polarity >= 0) - spec->gpio_led_polarity = default_polarity; - else - spec->gpio_led_polarity = 1; - return 1; + /* power down inactive ADCs */ + if (spec->powerdown_adcs) { + for (i = 0; i < spec->gen.num_all_adcs; i++) { + if (spec->active_adcs & (1 << i)) + continue; + snd_hda_codec_write(codec, spec->gen.all_adcs[i], 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); } } - /* - * Fallback case - if we don't find the DMI strings, - * we statically set the GPIO - if not a B-series system - * and default polarity is provided - */ - if (!hp_blike_system(codec->subsystem_id) && - (default_polarity == 0 || default_polarity == 1)) { - set_hp_led_gpio(codec); - spec->gpio_led_polarity = default_polarity; - return 1; + /* power down unused DACs */ + for (i = 0; i < spec->gen.num_all_dacs; i++) { + if (!snd_hda_get_nid_path(codec, spec->gen.all_dacs[i], 0)) + snd_hda_codec_write(codec, spec->gen.all_dacs[i], 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); } + return 0; } -static int hp_blike_system(u32 subsystem_id) +static void stac_shutup(struct hda_codec *codec) { - switch (subsystem_id) { - case 0x103c1520: - case 0x103c1521: - case 0x103c1523: - case 0x103c1524: - case 0x103c1525: - case 0x103c1722: - case 0x103c1723: - case 0x103c1724: - case 0x103c1725: - case 0x103c1726: - case 0x103c1727: - case 0x103c1728: - case 0x103c1729: - case 0x103c172a: - case 0x103c172b: - case 0x103c307e: - case 0x103c307f: - case 0x103c3080: - case 0x103c3081: - case 0x103c7007: - case 0x103c7008: - return 1; - } - return 0; + struct sigmatel_spec *spec = codec->spec; + + snd_hda_shutup_pins(codec); + + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); +} + +static void stac_free(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!spec) + return; + + snd_hda_gen_spec_free(&spec->gen); + kfree(spec); + snd_hda_detach_beep_device(codec); } #ifdef CONFIG_PROC_FS @@ -6341,101 +3585,69 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #endif #ifdef CONFIG_PM -static int stac92xx_resume(struct hda_codec *codec) +static int stac_resume(struct hda_codec *codec) { - stac92xx_init(codec); + codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); - /* fake event to set up pins again to override cached values */ - stac_fake_hp_events(codec); return 0; } -static int stac92xx_suspend(struct hda_codec *codec) -{ - stac92xx_shutup(codec); - return 0; -} - -static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) -{ - unsigned int afg_power_state = power_state; - struct sigmatel_spec *spec = codec->spec; - - if (power_state == AC_PWRST_D3) { - if (spec->vref_mute_led_nid) { - /* with vref-out pin used for mute led control - * codec AFG is prevented from D3 state - */ - afg_power_state = AC_PWRST_D1; - } - /* this delay seems necessary to avoid click noise at power-down */ - msleep(100); - } - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - afg_power_state); - snd_hda_codec_set_power_to_all(codec, fg, power_state, true); -} -#else -#define stac92xx_suspend NULL -#define stac92xx_resume NULL -#define stac92xx_set_power_state NULL -#endif /* CONFIG_PM */ - -/* update mute-LED accoring to the master switch */ -static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) -{ - struct sigmatel_spec *spec = codec->spec; - int muted = !enabled; - - if (!spec->gpio_led) - return; - - /* LED state is inverted on these systems */ - if (spec->gpio_led_polarity) - muted = !muted; - - if (!spec->vref_mute_led_nid) { - if (muted) - spec->gpio_data |= spec->gpio_led; - else - spec->gpio_data &= ~spec->gpio_led; - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - } else { - spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; - stac_vrefout_set(codec, spec->vref_mute_led_nid, - spec->vref_led); +static int stac_suspend(struct hda_codec *codec) +{ + stac_shutup(codec); + return 0; +} + +static void stac_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + unsigned int afg_power_state = power_state; + struct sigmatel_spec *spec = codec->spec; + + if (power_state == AC_PWRST_D3) { + if (spec->vref_mute_led_nid) { + /* with vref-out pin used for mute led control + * codec AFG is prevented from D3 state + */ + afg_power_state = AC_PWRST_D1; + } + /* this delay seems necessary to avoid click noise at power-down */ + msleep(100); } + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, + afg_power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } +#else +#define stac_suspend NULL +#define stac_resume NULL +#define stac_set_power_state NULL +#endif /* CONFIG_PM */ -static const struct hda_codec_ops stac92xx_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac92xx_init, - .free = stac92xx_free, +static const struct hda_codec_ops stac_patch_ops = { + .build_controls = snd_hda_gen_build_controls, + .build_pcms = snd_hda_gen_build_pcms, + .init = stac_init, + .free = stac_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM - .suspend = stac92xx_suspend, - .resume = stac92xx_resume, + .suspend = stac_suspend, + .resume = stac_resume, #endif - .reboot_notify = stac92xx_shutup, + .reboot_notify = stac_shutup, }; -static int alloc_stac_spec(struct hda_codec *codec, int num_pins, - const hda_nid_t *pin_nids) +static int alloc_stac_spec(struct hda_codec *codec) { struct sigmatel_spec *spec; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; codec->no_trigger_sense = 1; /* seems common with STAC/IDT codecs */ - spec->num_pins = num_pins; - spec->pin_nids = pin_nids; - snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32); return 0; } @@ -6444,40 +3656,28 @@ static int patch_stac9200(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac9200_pin_nids), - stac9200_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac9200_models, stac9200_fixup_tbl, - stac9200_fixups); + codec->patch_ops = stac_patch_ops; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = stac9200_dac_nids; - spec->adc_nids = stac9200_adc_nids; - spec->mux_nids = stac9200_mux_nids; - spec->num_muxes = 1; - spec->num_dmics = 0; - spec->num_adcs = 1; - spec->num_pwrs = 0; snd_hda_add_verbs(codec, stac9200_eapd_init); - spec->mixer = stac9200_mixer; - + snd_hda_pick_fixup(codec, stac9200_models, stac9200_fixup_tbl, + stac9200_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac9200_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6488,56 +3688,28 @@ static int patch_stac925x(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac925x_pin_nids), - stac925x_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac925x_models, stac925x_fixup_tbl, - stac925x_fixups); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = stac925x_dac_nids; - spec->adc_nids = stac925x_adc_nids; - spec->mux_nids = stac925x_mux_nids; - spec->num_muxes = 1; - spec->num_adcs = 1; - spec->num_pwrs = 0; - switch (codec->vendor_id) { - case 0x83847632: /* STAC9202 */ - case 0x83847633: /* STAC9202D */ - case 0x83847636: /* STAC9251 */ - case 0x83847637: /* STAC9251D */ - spec->num_dmics = STAC925X_NUM_DMICS; - spec->dmic_nids = stac925x_dmic_nids; - spec->num_dmuxes = ARRAY_SIZE(stac925x_dmux_nids); - spec->dmux_nids = stac925x_dmux_nids; - break; - default: - spec->num_dmics = 0; - break; - } + codec->patch_ops = stac_patch_ops; snd_hda_add_verbs(codec, stac925x_core_init); - spec->mixer = stac925x_mixer; - spec->num_caps = 1; - spec->capvols = stac925x_capvols; - spec->capsws = stac925x_capsws; + snd_hda_pick_fixup(codec, stac925x_models, stac925x_fixup_tbl, + stac925x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -6546,86 +3718,66 @@ static int patch_stac925x(struct hda_codec *codec) static int patch_stac92hd73xx(struct hda_codec *codec) { struct sigmatel_spec *spec; - hda_nid_t conn[STAC92HD73_DAC_COUNT + 2]; int err; int num_dacs; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac92hd73xx_pin_nids), - stac92hd73xx_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 0; - codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; - - snd_hda_pick_fixup(codec, stac92hd73xx_models, stac92hd73xx_fixup_tbl, - stac92hd73xx_fixups); - - num_dacs = snd_hda_get_connections(codec, 0x0a, - conn, STAC92HD73_DAC_COUNT + 2) - 1; + num_dacs = snd_hda_get_num_conns(codec, 0x0a) - 1; if (num_dacs < 3 || num_dacs > 5) { printk(KERN_WARNING "hda_codec: Could not determine " "number of channels defaulting to DAC count\n"); - num_dacs = STAC92HD73_DAC_COUNT; + num_dacs = 5; } switch (num_dacs) { case 0x3: /* 6 Channel */ - spec->aloopback_ctl = stac92hd73xx_6ch_loopback; + spec->aloopback_ctl = &stac92hd73xx_6ch_loopback; break; case 0x4: /* 8 Channel */ - spec->aloopback_ctl = stac92hd73xx_8ch_loopback; + spec->aloopback_ctl = &stac92hd73xx_8ch_loopback; break; case 0x5: /* 10 Channel */ - spec->aloopback_ctl = stac92hd73xx_10ch_loopback; + spec->aloopback_ctl = &stac92hd73xx_10ch_loopback; break; } - spec->multiout.dac_nids = spec->dac_nids; spec->aloopback_mask = 0x01; spec->aloopback_shift = 8; spec->digbeep_nid = 0x1c; - spec->mux_nids = stac92hd73xx_mux_nids; - spec->adc_nids = stac92hd73xx_adc_nids; - spec->dmic_nids = stac92hd73xx_dmic_nids; - spec->dmux_nids = stac92hd73xx_dmux_nids; - spec->smux_nids = stac92hd73xx_smux_nids; - - spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); - - spec->num_caps = STAC92HD73XX_NUM_CAPS; - spec->capvols = stac92hd73xx_capvols; - spec->capsws = stac92hd73xx_capsws; /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; - spec->num_dmics = STAC92HD73XX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); spec->eapd_switch = 1; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; + spec->gen.own_eapd_ctl = 1; + + codec->patch_ops = stac_patch_ops; + + snd_hda_pick_fixup(codec, stac92hd73xx_models, stac92hd73xx_fixup_tbl, + stac92hd73xx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); if (!spec->volknob_init) snd_hda_add_verbs(codec, stac92hd73xx_core_init); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac92hd7x_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -6633,145 +3785,6 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return 0; } -static int hp_bnb2011_with_dock(struct hda_codec *codec) -{ - if (codec->vendor_id != 0x111d7605 && - codec->vendor_id != 0x111d76d1) - return 0; - - switch (codec->subsystem_id) { - case 0x103c1618: - case 0x103c1619: - case 0x103c161a: - case 0x103c161b: - case 0x103c161c: - case 0x103c161d: - case 0x103c161e: - case 0x103c161f: - - case 0x103c162a: - case 0x103c162b: - - case 0x103c1630: - case 0x103c1631: - - case 0x103c1633: - case 0x103c1634: - case 0x103c1635: - - case 0x103c3587: - case 0x103c3588: - case 0x103c3589: - case 0x103c358a: - - case 0x103c3667: - case 0x103c3668: - case 0x103c3669: - - return 1; - } - return 0; -} - -static void stac92hd8x_add_pin(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); - int i; - - spec->auto_pin_nids[spec->auto_pin_cnt] = nid; - spec->auto_pin_cnt++; - - if (get_defcfg_device(def_conf) == AC_JACK_MIC_IN && - get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) { - for (i = 0; i < ARRAY_SIZE(stac92hd83xxx_dmic_nids); i++) { - if (nid == stac92hd83xxx_dmic_nids[i]) { - spec->auto_dmic_nids[spec->auto_dmic_cnt] = nid; - spec->auto_dmic_cnt++; - } - } - } -} - -static void stac92hd8x_add_adc(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - - spec->auto_adc_nids[spec->auto_adc_cnt] = nid; - spec->auto_adc_cnt++; -} - -static void stac92hd8x_add_mux(struct hda_codec *codec, hda_nid_t nid) -{ - int i, j; - struct sigmatel_spec *spec = codec->spec; - - for (i = 0; i < spec->auto_adc_cnt; i++) { - if (get_connection_index(codec, - spec->auto_adc_nids[i], nid) >= 0) { - /* mux and volume for adc_nids[i] */ - if (!spec->auto_mux_nids[i]) { - spec->auto_mux_nids[i] = nid; - /* 92hd codecs capture volume is in mux */ - spec->auto_capvols[i] = HDA_COMPOSE_AMP_VAL(nid, - 3, 0, HDA_OUTPUT); - } - for (j = 0; j < spec->auto_dmic_cnt; j++) { - if (get_connection_index(codec, nid, - spec->auto_dmic_nids[j]) >= 0) { - /* dmux for adc_nids[i] */ - if (!spec->auto_dmux_nids[i]) - spec->auto_dmux_nids[i] = nid; - break; - } - } - break; - } - } -} - -static void stac92hd8x_fill_auto_spec(struct hda_codec *codec) -{ - hda_nid_t nid, end_nid; - unsigned int wid_caps, wid_type; - struct sigmatel_spec *spec = codec->spec; - - end_nid = codec->start_nid + codec->num_nodes; - - for (nid = codec->start_nid; nid < end_nid; nid++) { - wid_caps = get_wcaps(codec, nid); - wid_type = get_wcaps_type(wid_caps); - - if (wid_type == AC_WID_PIN) - stac92hd8x_add_pin(codec, nid); - - if (wid_type == AC_WID_AUD_IN && !(wid_caps & AC_WCAP_DIGITAL)) - stac92hd8x_add_adc(codec, nid); - } - - for (nid = codec->start_nid; nid < end_nid; nid++) { - wid_caps = get_wcaps(codec, nid); - wid_type = get_wcaps_type(wid_caps); - - if (wid_type == AC_WID_AUD_SEL) - stac92hd8x_add_mux(codec, nid); - } - - spec->pin_nids = spec->auto_pin_nids; - spec->num_pins = spec->auto_pin_cnt; - spec->adc_nids = spec->auto_adc_nids; - spec->num_adcs = spec->auto_adc_cnt; - spec->capvols = spec->auto_capvols; - spec->capsws = spec->auto_capvols; - spec->num_caps = spec->auto_adc_cnt; - spec->mux_nids = spec->auto_mux_nids; - spec->num_muxes = spec->auto_adc_cnt; - spec->dmux_nids = spec->auto_dmux_nids; - spec->num_dmuxes = spec->auto_adc_cnt; - spec->dmic_nids = spec->auto_dmic_nids; - spec->num_dmics = spec->auto_dmic_cnt; -} - static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -6783,7 +3796,7 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_data |= spec->gpio_led; } else { codec->patch_ops.set_power_state = - stac92xx_set_power_state; + stac_set_power_state; } } @@ -6792,6 +3805,8 @@ static void stac_setup_gpio(struct hda_codec *codec) spec->gpio_dir |= spec->mic_mute_led_gpio; spec->mic_mute_led_on = true; spec->gpio_data |= spec->mic_mute_led_gpio; + + spec->gen.capture_switch_hook = stac_capture_led_hook; } } @@ -6800,36 +3815,34 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, 0, NULL); /* pins filled later */ + err = alloc_stac_spec(codec); if (err < 0) return err; codec->epss = 0; /* longer delay needed for D3 */ - stac92hd8x_fill_auto_spec(codec); spec = codec->spec; spec->linear_tone_beep = 0; - codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; + spec->gen.own_eapd_ctl = 1; + spec->digbeep_nid = 0x21; spec->pwr_nids = stac92hd83xxx_pwr_nids; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); - spec->multiout.dac_nids = spec->dac_nids; + spec->default_polarity = -1; /* no default cfg */ - snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, - stac92hd83xxx_fixups); + codec->patch_ops = stac_patch_ops; snd_hda_add_verbs(codec, stac92hd83xxx_core_init); - spec->default_polarity = -1; /* no default cfg */ - - codec->patch_ops = stac92xx_patch_ops; + snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, + stac92hd83xxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); stac_setup_gpio(codec); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } @@ -6840,133 +3853,31 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) return 0; } -static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, - hda_nid_t dig0pin) -{ - struct sigmatel_spec *spec = codec->spec; - int idx; - - for (idx = 0; idx < spec->num_pins; idx++) - if (spec->pin_nids[idx] == dig0pin) - break; - if ((idx + 2) >= spec->num_pins) - return 0; - - /* dig1pin case */ - if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE) - return 2; - - /* dig0pin + dig2pin case */ - if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE) - return 2; - if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE) - return 1; - else - return 0; -} - -/* HP dv7 bass switch - GPIO5 */ -#define stac_hp_bass_gpio_info snd_ctl_boolean_mono_info -static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20); - return 0; -} - -static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - unsigned int gpio_data; - - gpio_data = (spec->gpio_data & ~0x20) | - (ucontrol->value.integer.value[0] ? 0x20 : 0); - if (gpio_data == spec->gpio_data) - return 0; - spec->gpio_data = gpio_data; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); - return 1; -} - -static const struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = stac_hp_bass_gpio_info, - .get = stac_hp_bass_gpio_get, - .put = stac_hp_bass_gpio_put, -}; - -static int stac_add_hp_bass_switch(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - - if (!stac_control_new(spec, &stac_hp_bass_sw_ctrl, - "Bass Speaker Playback Switch", 0)) - return -ENOMEM; - - spec->gpio_mask |= 0x20; - spec->gpio_dir |= 0x20; - spec->gpio_data |= 0x20; - return 0; -} - static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; const struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err; - err = alloc_stac_spec(codec, STAC92HD71BXX_NUM_PINS, - stac92hd71bxx_pin_nids_4port); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 0; - codec->patch_ops = stac92xx_patch_ops; - switch (codec->vendor_id) { - case 0x111d76b6: - case 0x111d76b7: - break; - case 0x111d7603: - case 0x111d7608: - /* On 92HD75Bx 0x27 isn't a pin nid */ - spec->num_pins--; - /* fallthrough */ - default: - spec->pin_nids = stac92hd71bxx_pin_nids_6port; - } - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac92hd71bxx_models, stac92hd71bxx_fixup_tbl, - stac92hd71bxx_fixups); + codec->patch_ops = stac_patch_ops; /* GPIO0 = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; spec->gpio_data = 0x01; - spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nids = stac92hd71bxx_dmux_nids; - - spec->num_caps = STAC92HD71BXX_NUM_CAPS; - spec->capvols = stac92hd71bxx_capvols; - spec->capsws = stac92hd71bxx_capsws; - switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: unmute_init++; - /* fallthru */ - case 0x111d76b4: /* 6 Port without Analog Mixer */ - case 0x111d76b5: - codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd71bxx_dmic_nids, - STAC92HD71BXX_NUM_DMICS); break; case 0x111d7608: /* 5 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 0 || @@ -6977,21 +3888,11 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); - spec->dmic_nids = stac92hd71bxx_dmic_5port_nids; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd71bxx_dmic_5port_nids, - STAC92HD71BXX_NUM_DMICS - 1); break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) spec->stream_delay = 40; /* 40 milliseconds */ - /* fallthru */ - default: - codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd71bxx_dmic_nids, - STAC92HD71BXX_NUM_DMICS); break; } @@ -7001,32 +3902,24 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); - spec->aloopback_ctl = stac92hd71bxx_loopback; + spec->aloopback_ctl = &stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; spec->powerdown_adcs = 1; spec->digbeep_nid = 0x26; - spec->mux_nids = stac92hd71bxx_mux_nids; - spec->adc_nids = stac92hd71bxx_adc_nids; - spec->smux_nids = stac92hd71bxx_smux_nids; + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; - spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); - - spec->multiout.dac_nids = spec->dac_nids; - spec->default_polarity = 1; - + snd_hda_pick_fixup(codec, stac92hd71bxx_models, stac92hd71bxx_fixup_tbl, + stac92hd71bxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); stac_setup_gpio(codec); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } @@ -7042,42 +3935,18 @@ static int patch_stac922x(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac922x_pin_nids), - stac922x_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac922x_models, stac922x_fixup_tbl, - stac922x_fixups); - - spec->adc_nids = stac922x_adc_nids; - spec->mux_nids = stac922x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac922x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac922x_adc_nids); - spec->num_dmics = 0; - spec->num_pwrs = 0; + codec->patch_ops = stac_patch_ops; - spec->num_caps = STAC922X_NUM_CAPS; - spec->capvols = stac922x_capvols; - spec->capsws = stac922x_capsws; - - spec->multiout.dac_nids = spec->dac_nids; - snd_hda_add_verbs(codec, stac922x_core_init); - snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - - err = stac92xx_parse_auto_config(codec); - if (err < 0) { - stac92xx_free(codec); - return err; - } - - codec->patch_ops = stac92xx_patch_ops; - /* Fix Mux capture level; max to 2 */ snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, (0 << AC_AMPCAP_OFFSET_SHIFT) | @@ -7085,6 +3954,16 @@ static int patch_stac922x(struct hda_codec *codec) (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | (0 << AC_AMPCAP_MUTE_SHIFT)); + snd_hda_pick_fixup(codec, stac922x_models, stac922x_fixup_tbl, + stac922x_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + + err = stac_parse_auto_config(codec); + if (err < 0) { + stac_free(codec); + return err; + } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); return 0; @@ -7095,58 +3974,40 @@ static int patch_stac927x(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac927x_pin_nids), - stac927x_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; - codec->slave_dig_outs = stac927x_slave_dig_outs; - - snd_hda_pick_fixup(codec, stac927x_models, stac927x_fixup_tbl, - stac927x_fixups); + spec->gen.own_eapd_ctl = 1; spec->digbeep_nid = 0x23; - spec->adc_nids = stac927x_adc_nids; - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->smux_nids = stac927x_smux_nids; - spec->num_smuxes = ARRAY_SIZE(stac927x_smux_nids); - spec->spdif_labels = stac927x_spdif_labels; - spec->dac_list = stac927x_dac_nids; - spec->multiout.dac_nids = spec->dac_nids; /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = 0x01; spec->gpio_dir = spec->gpio_data = 0x01; - spec->num_dmics = 0; - - spec->num_caps = STAC927X_NUM_CAPS; - spec->capvols = stac927x_capvols; - spec->capsws = stac927x_capsws; - - spec->num_pwrs = 0; - spec->aloopback_ctl = stac927x_loopback; + spec->aloopback_ctl = &stac927x_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; spec->eapd_switch = 1; + codec->patch_ops = stac_patch_ops; + + snd_hda_pick_fixup(codec, stac927x_models, stac927x_fixup_tbl, + stac927x_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); if (!spec->volknob_init) snd_hda_add_verbs(codec, stac927x_core_init); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac927x_proc_hook; /* @@ -7171,40 +4032,21 @@ static int patch_stac9205(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac9205_pin_nids), - stac9205_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; - - snd_hda_pick_fixup(codec, stac9205_models, stac9205_fixup_tbl, - stac9205_fixups); + spec->gen.own_eapd_ctl = 1; spec->digbeep_nid = 0x23; - spec->adc_nids = stac9205_adc_nids; - spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids); - spec->mux_nids = stac9205_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); - spec->smux_nids = stac9205_smux_nids; - spec->num_smuxes = ARRAY_SIZE(stac9205_smux_nids); - spec->dmic_nids = stac9205_dmic_nids; - spec->num_dmics = STAC9205_NUM_DMICS; - spec->dmux_nids = stac9205_dmux_nids; - spec->num_dmuxes = ARRAY_SIZE(stac9205_dmux_nids); - spec->num_pwrs = 0; snd_hda_add_verbs(codec, stac9205_core_init); - spec->aloopback_ctl = stac9205_loopback; - - spec->num_caps = STAC9205_NUM_CAPS; - spec->capvols = stac9205_capvols; - spec->capsws = stac9205_capsws; + spec->aloopback_ctl = &stac9205_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; - spec->multiout.dac_nids = spec->dac_nids; /* GPIO0 High = EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; @@ -7213,16 +4055,18 @@ static int patch_stac9205(struct hda_codec *codec) /* Turn on/off EAPD per HP plugging */ spec->eapd_switch = 1; + codec->patch_ops = stac_patch_ops; + + snd_hda_pick_fixup(codec, stac9205_models, stac9205_fixup_tbl, + stac9205_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return err; } - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac9205_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); @@ -7240,24 +4084,6 @@ static const struct hda_verb stac9872_core_init[] = { {} }; -static const hda_nid_t stac9872_pin_nids[] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, - 0x11, 0x13, 0x14, -}; - -static const hda_nid_t stac9872_adc_nids[] = { - 0x8 /*,0x6*/ -}; - -static const hda_nid_t stac9872_mux_nids[] = { - 0x15 -}; - -static const unsigned long stac9872_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), -}; -#define stac9872_capsws stac9872_capvols - static const struct hda_pintbl stac9872_vaio_pin_configs[] = { { 0x0a, 0x03211020 }, { 0x0b, 0x411111f0 }, @@ -7294,36 +4120,27 @@ static int patch_stac9872(struct hda_codec *codec) struct sigmatel_spec *spec; int err; - err = alloc_stac_spec(codec, ARRAY_SIZE(stac9872_pin_nids), - stac9872_pin_nids); + err = alloc_stac_spec(codec); if (err < 0) return err; spec = codec->spec; spec->linear_tone_beep = 1; + spec->gen.own_eapd_ctl = 1; - snd_hda_pick_fixup(codec, stac9872_models, stac9872_fixup_tbl, - stac9872_fixups); + codec->patch_ops = stac_patch_ops; - spec->multiout.dac_nids = spec->dac_nids; - spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); - spec->adc_nids = stac9872_adc_nids; - spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); - spec->mux_nids = stac9872_mux_nids; - spec->num_caps = 1; - spec->capvols = stac9872_capvols; - spec->capsws = stac9872_capsws; snd_hda_add_verbs(codec, stac9872_core_init); + snd_hda_pick_fixup(codec, stac9872_models, stac9872_fixup_tbl, + stac9872_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = stac92xx_parse_auto_config(codec); + err = stac_parse_auto_config(codec); if (err < 0) { - stac92xx_free(codec); + stac_free(codec); return -EINVAL; } - spec->input_mux = &spec->private_imux; - codec->patch_ops = stac92xx_patch_ops; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); -- 1.8.1

1 0

[alsa-devel] [PATCH RFC 0/9] Final bits for HD-audio generic parser
by Takashi Iwai 17 Jan '13

17 Jan '13

Hi, this is the final bits for improving / fixing the HD-audio generic parser for migrating patch_sigmatel.c. After these, patch_sigmatel.c is converted to use the generic parser, and all done! As usual, the patches are found in sound-unstable git tree test/hda-gen-parser branch (it's already merged into master and test/hda-migrate branch, too). Takashi

1 9

[alsa-devel] [PATCH 01/19] ALSA: add Yamaha YMU831 codec support
by Yoichi Yuasa 17 Jan '13

17 Jan '13

Signed-off-by: Yoichi Yuasa <yuasa(a)linux-mips.org> --- sound/soc/codecs/Kconfig | 6 ++++++ sound/soc/codecs/Makefile | 1 + sound/soc/codecs/ymu831/Makefile | 3 +++ 3 files changed, 10 insertions(+) create mode 100644 sound/soc/codecs/ymu831/Makefile diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a84782..bb405b87 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS + select SND_SOC_YMU831 if SPI_MASTER help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine @@ -494,6 +495,11 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate +config SND_SOC_YMU831 + tristate + select SND_HWDEP + + # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f6e8e36..1d31783 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -236,6 +236,7 @@ obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_ADSP) += snd-soc-wm-adsp.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o +obj-$(CONFIG_SND_SOC_YMU831) += ymu831/ # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/ymu831/Makefile b/sound/soc/codecs/ymu831/Makefile new file mode 100644 index 0000000..5b3f66d --- /dev/null +++ b/sound/soc/codecs/ymu831/Makefile @@ -0,0 +1,3 @@ +snd-soc-ymu831-objs := \ + +obj-$(CONFIG_SND_SOC_YMU831) += snd-soc-ymu831.o -- 1.7.9.5

3 41

[alsa-devel] [PATCH v2 0/3] Tegra AC97 v2
by Lucas Stach 17 Jan '13

17 Jan '13

v2 of the Tegra AC97 series. Unmerged ASoC related patches only. The machine driver now instantiates the codec driver by itself, without usage of DT. Other fixups for review feedback also included. I'll post a patch for the AC97 clock once Prashant patches land in a branch. Lucas Stach (3): ASoC: tegra: add ac97 host driver ASoC: tegra: add ac97 host controller to device tree ASoC: tegra: add tegra machine driver using wm9712 codec .../bindings/sound/nvidia,tegra-audio-wm9712.txt | 51 +++ .../bindings/sound/nvidia,tegra20-ac97.txt | 22 + arch/arm/boot/dts/tegra20.dtsi | 9 + sound/soc/tegra/Kconfig | 19 + sound/soc/tegra/Makefile | 4 + sound/soc/tegra/tegra20_ac97.c | 480 +++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.h | 95 ++++ sound/soc/tegra/tegra_wm9712.c | 179 ++++++++ 8 files changed, 859 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra20-ac97.txt create mode 100644 sound/soc/tegra/tegra20_ac97.c create mode 100644 sound/soc/tegra/tegra20_ac97.h create mode 100644 sound/soc/tegra/tegra_wm9712.c -- 1.8.0.1

3 13

[alsa-devel] [PATCH alsa-utils] speaker-test: increase the maximum supported rate to 384000
by Antonio Ospite 17 Jan '13

17 Jan '13

There are some devices around supporting this sample rate so let's make speaker-test capable to deal with them. Signed-off-by: Antonio Ospite <ao2(a)amarulasolutions.com> --- speaker-test/speaker-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 3c89c38..de42a0c 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -999,7 +999,7 @@ int main(int argc, char *argv[]) { case 'r': rate = atoi(optarg); rate = rate < 4000 ? 4000 : rate; - rate = rate > 196000 ? 196000 : rate; + rate = rate > 384000 ? 384000 : rate; break; case 'c': channels = atoi(optarg); -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: arizona: Use actual rather than desired BCLK when calculating LRCLK
by Mark Brown 17 Jan '13

17 Jan '13

Otherwise we'll get the wrong LRCLK if we need to pick a higher BCLK than is required. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index c3592db..ef62c43 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -685,7 +685,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, } sr_val = i; - lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + lrclk = rates[bclk] / params_rate(params); arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", rates[bclk], rates[bclk] / lrclk); -- 1.7.10.4

1 0

[alsa-devel] [PATCH v3 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
by mengdong.lin＠intel.com 17 Jan '13

17 Jan '13

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI or Display Port. So this patch adds fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the pins' connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..59abe73 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1687,6 +1687,30 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + /* override 3 pins connection list */ + snd_hda_override_conn_list(codec, 0x05, 3, list); + snd_hda_override_conn_list(codec, 0x06, 3, list); + snd_hda_override_conn_list(codec, 0x07, 3, list); +} + + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,6 +1720,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + + if (codec->vendor_id == 0x80862807) + intel_haswell_fixup_connect_list(codec); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.7.9.5

4 5

[alsa-devel] [PATCH 0/3] ...and three more fixes for the new generic parser
by David Henningsson 16 Jan '13

16 Jan '13

These regressions were found when running the test tool over even more alsa-infos. Btw, would you mind updating the hda-migrate branch too? It's useful to have your latest fixes from the hda-gen-parser branch when I'm running pre-release QA testing. David Henningsson (3): ALSA: hda - do not add non-existing Mic boost controls ALSA: hda - force different capture controls if amp caps differ ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs sound/pci/hda/hda_generic.c | 31 ++++++++++++++++++++++++++----- 1 file changed, 26 insertions(+), 5 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] alsa-util fix (Re: [PATCH] alsa-lib: Fix build with automake-1.13)
by Marko Lindqvist 16 Jan '13

16 Jan '13

On 4 January 2013 01:40, Marko Lindqvist <cazfi74(a)gmail.com> wrote: > Automake-1.13 removed long obsolete AM_CONFIG_HEADER completely ( > http://lists.gnu.org/archive/html/automake/2012-12/msg00038.html ) and > errors out upon seeing it. > > Attached patch replaces it with proper AC_CONFIG_HEADERS. > > > I'm not subscribed to the list. I'm mass-submitting automake-1.13 fixes. Please see what you can do so I don't need to do the same when automake-1.14 comes out: http://cazfi.livejournal.com/195108.html - ML

2 1

[alsa-devel] [PATCH] ASoC: fsi: Remove __devinitconst
by Thierry Reding 16 Jan '13

16 Jan '13

__devinitconst and friends have recently been removed and must not be used anymore. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9157612..c724026 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2160,7 +2160,7 @@ static struct fsi_core fsi2_core = { .b_mclk = B_MST_CTLR, }; -static struct of_device_id fsi_of_match[] __devinitconst = { +static struct of_device_id fsi_of_match[] = { { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, {}, -- 1.8.1

2 1

[alsa-devel] [PATCH] ASoC: wm8978: SND_SOC_DAIFMT_NB_NF become default clock inversion
by Kuninori Morimoto 16 Jan '13

16 Jan '13

Current wm8978 driver return -EINVAL if it doesn't have SND_SOC_DAIFMT_INV_MASK flags on snd_soc_dai_ops :: set_fmt. But default settings should be treated as SND_SOC_DAIFMT_NB_NF (= no clock inversion). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/wm8978.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index f347af3..ba01981 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -692,8 +692,6 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; case SND_SOC_DAIFMT_IB_IF: iface |= 0x180; break; @@ -703,8 +701,9 @@ static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_IF: iface |= 0x80; break; + case SND_SOC_DAIFMT_NB_NF: default: - return -EINVAL; + break; } snd_soc_write(codec, WM8978_AUDIO_INTERFACE, iface); -- 1.7.9.5

2 3

[alsa-devel] [PATCH 0/3] Fixes for the new generic parser
by David Henningsson 16 Jan '13

16 Jan '13

Hi Takashi, I've run my hda-emu summary test tool to check for errors in the 110 codecs we have of certified systems (those in codecs/canonical) against the test/hda-migrate branch. After these fixes and the one I sent yesterday, there are no remaining regressions (as seen by this test tool) compared to the current sound tree. David Henningsson (3): ALSA: hda - Check array bounds in get_input_path ALSA: hda - fix wrong adc_idx in generic parser ALSA: hda - initialize channel counts correctly sound/pci/hda/hda_generic.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: arizona: Don't request FLL lock IRQ
by Mark Brown 16 Jan '13

16 Jan '13

We only log the result and since the interrupt triggers on loss of lock during shutdown this may lead to spurious interrupts during shutdown delaying the process. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 19 ------------------- sound/soc/codecs/arizona.h | 1 - 2 files changed, 20 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9da5eee..b3621d6 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1008,17 +1008,6 @@ int arizona_init_dai(struct arizona_priv *priv, int id) } EXPORT_SYMBOL_GPL(arizona_init_dai); -static irqreturn_t arizona_fll_lock(int irq, void *data) -{ - struct arizona_fll *fll = data; - - arizona_fll_dbg(fll, "Lock status changed\n"); - - complete(&fll->lock); - - return IRQ_HANDLED; -} - static irqreturn_t arizona_fll_clock_ok(int irq, void *data) { struct arizona_fll *fll = data; @@ -1258,7 +1247,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, { int ret; - init_completion(&fll->lock); init_completion(&fll->ok); fll->id = id; @@ -1269,13 +1257,6 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); - ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, - arizona_fll_lock, fll); - if (ret != 0) { - dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", - id, ret); - } - ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, arizona_fll_clock_ok, fll); if (ret != 0) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b3b97f56..67fd49b 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -200,7 +200,6 @@ struct arizona_fll { int id; unsigned int base; unsigned int vco_mult; - struct completion lock; struct completion ok; unsigned int fref; unsigned int fout; -- 1.7.10.4

1 0

[alsa-devel] Fujitsu Lifebook s7110 (ALC262) no sound
by Raymond Yau 16 Jan '13

16 Jan '13

> after patching the clean 12.04 install with teh last alsa version (see comment #13) and with external speakers plugged (3.5 mm jack), I have got the follwoing Alsa log: http://www.alsa-project.org/db/?f=778415ad65d437d9ff5e5dd341e0e556ff848bc1 http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=commitdiff;h=3d… -/* - * fujitsu model - * 0x14 = headphone/spdif-out, 0x15 = internal speaker, - * 0x1b = port replicator headphone out - */ Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Surround Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Surround Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0x032110f0: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=03, enabled=1 Connection: 2 0x0c 0x0d* Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Front Phantom Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003e: IN OUT HP Detect Trigger Pin Default 0xb7231110: [Fixed] HP Out at Oth Mobile-In Conn = ATAPI, Color = Black DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Dock Headphone Jack", index=0, device=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x2321101f: [Jack] HP Out at Sep Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 2 0x0c* 0x0d If seem that BIOS did not setup the internal speaker node and need pin fixup Pulseaudio expect headphone Jack and speaker phantom port https://launchpadlibrarian.net/128561206/pulseverbose2.log

2 1

[alsa-devel] [PATCH v3 0/4] RME HDSP fixes
by Adrian Knoth 16 Jan '13

16 Jan '13

Hi! Changes compared to v2: - use snd_printd to print iobox detection time - use KERN_ERR in case no iobox was found - define HDSP_S300 as a dummy for (0x100 | HDSP_S200) Adrian Knoth (4): ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes ALSA: hdsp - Implement generic function to toggle settings ALSA: hdsp - Use HDSP_TOGGLE_SETTING to alter settings ALSA: hdsp - Remove obsolete settings functions sound/pci/rme9652/hdsp.c | 462 ++++++++++++---------------------------------- 1 file changed, 117 insertions(+), 345 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH] ASoC: maintainers: Update email address.
by Liam Girdwood 16 Jan '13

16 Jan '13

From: Liam Girdwood <liam.r.girdwood(a)intel.com> I'm no longer at TI. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)intel.com> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 915564e..3c2e275 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7082,7 +7082,7 @@ F: include/uapi/sound/ F: sound/ SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) -M: Liam Girdwood <lrg(a)ti.com> +M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -- 1.7.10.4

2 1

[alsa-devel] [PATCH v2 0/4] RME HDSP fixes
by Adrian Knoth 15 Jan '13

15 Jan '13

Hi, here's v2 of my previous series of patches for the RME HDSP. Changes to the previous version: - correct time value in verbose output ("iobox found after %d ms") - define and use constants for iobox detection (no more 0x10a0) - hdsp_toggle_setting: only change register if newval != oldval Adrian Knoth (4): ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes ALSA: hdsp - Implement generic function to toggle settings ALSA: hdsp - Use HDSP_TOGGLE_SETTING to alter settings ALSA: hdsp - Remove obsolete settings functions sound/pci/rme9652/hdsp.c | 461 ++++++++++++---------------------------------- 1 file changed, 116 insertions(+), 345 deletions(-) -- 1.7.10.4

2 5

[alsa-devel] [PATCH (sound-unstable)] ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
by David Henningsson 15 Jan '13

15 Jan '13

Obvious copy-paste error. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b28e403..e6cdad7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3794,7 +3794,7 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) } for (i = 0; i < codec->amp_cache.buf.used; i++) { struct hda_amp_info *amp; - amp = snd_array_elem(&codec->cmd_cache.buf, i); + amp = snd_array_elem(&codec->amp_cache.buf, i); amp->head.dirty = 1; } } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Make it work with sound-unstable.git
by David Henningsson 15 Jan '13

15 Jan '13

To be able to build the new code (so we can run aggregate tests on it) the following patch needs to be applied. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 8 +++++++ hda-ctlsh.c | 5 ++++ include/linux/ctype.h | 64 +++++++++++++++++++++++++++++++++++++++++++++++++ snd-wrapper.c | 32 +++++++++++++++++++++++++ 4 files changed, 109 insertions(+) create mode 100644 include/linux/ctype.h diff --git a/configure.ac b/configure.ac index 3103eb5..80b68a6 100644 --- a/configure.ac +++ b/configure.ac @@ -182,6 +182,14 @@ if test "$OLD_HDA_CMD" = "1"; then AC_DEFINE(OLD_HDA_CMD) fi +AC_MSG_CHECKING(for user_mutex) +if grep -q 'struct mutex user_mutex' hda/hda_codec.h; then + AC_DEFINE(HAVE_USER_MUTEX) + AC_MSG_RESULT(yes) +else + AC_MSG_RESULT(no) +fi + AC_MSG_CHECKING(for new pm_notify callback) ac_save_CFLAGS="$CFLAGS" CFLAGS="$CFLAGS -Iinclude" diff --git a/hda-ctlsh.c b/hda-ctlsh.c index 5ad95fc..04fc318 100644 --- a/hda-ctlsh.c +++ b/hda-ctlsh.c @@ -537,6 +537,11 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) return -ENOENT; } +int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) +{ + return -ENOENT; +} + #ifdef HAVE_USER_PINCFGS static void get_drv_pincfgs(void) { diff --git a/include/linux/ctype.h b/include/linux/ctype.h new file mode 100644 index 0000000..8acfe31 --- /dev/null +++ b/include/linux/ctype.h @@ -0,0 +1,64 @@ +#ifndef _LINUX_CTYPE_H +#define _LINUX_CTYPE_H + +/* + * NOTE! This ctype does not handle EOF like the standard C + * library is required to. + */ + +#define _U 0x01 /* upper */ +#define _L 0x02 /* lower */ +#define _D 0x04 /* digit */ +#define _C 0x08 /* cntrl */ +#define _P 0x10 /* punct */ +#define _S 0x20 /* white space (space/lf/tab) */ +#define _X 0x40 /* hex digit */ +#define _SP 0x80 /* hard space (0x20) */ + +extern const unsigned char _ctype[]; + +#define __ismask(x) (_ctype[(int)(unsigned char)(x)]) + +#define isalnum(c) ((__ismask(c)&(_U|_L|_D)) != 0) +#define isalpha(c) ((__ismask(c)&(_U|_L)) != 0) +#define iscntrl(c) ((__ismask(c)&(_C)) != 0) +#define isdigit(c) ((__ismask(c)&(_D)) != 0) +#define isgraph(c) ((__ismask(c)&(_P|_U|_L|_D)) != 0) +#define islower(c) ((__ismask(c)&(_L)) != 0) +#define isprint(c) ((__ismask(c)&(_P|_U|_L|_D|_SP)) != 0) +#define ispunct(c) ((__ismask(c)&(_P)) != 0) +/* Note: isspace() must return false for %NUL-terminator */ +#define isspace(c) ((__ismask(c)&(_S)) != 0) +#define isupper(c) ((__ismask(c)&(_U)) != 0) +#define isxdigit(c) ((__ismask(c)&(_D|_X)) != 0) + +#define isascii(c) (((unsigned char)(c))<=0x7f) +#define toascii(c) (((unsigned char)(c))&0x7f) + +static inline unsigned char __tolower(unsigned char c) +{ + if (isupper(c)) + c -= 'A'-'a'; + return c; +} + +static inline unsigned char __toupper(unsigned char c) +{ + if (islower(c)) + c -= 'a'-'A'; + return c; +} + +#define tolower(c) __tolower(c) +#define toupper(c) __toupper(c) + +/* + * Fast implementation of tolower() for internal usage. Do not use in your + * code. + */ +static inline char _tolower(const char c) +{ + return c | 0x20; +} + +#endif diff --git a/snd-wrapper.c b/snd-wrapper.c index 69cd19c..c2649d9 100644 --- a/snd-wrapper.c +++ b/snd-wrapper.c @@ -27,6 +27,35 @@ #include "hda/hda_codec.h" #include "hda-types.h" #include "hda-log.h" +#include <linux/ctype.h> + +/* Copied from lib/ctype.c */ +const unsigned char _ctype[] = { +_C,_C,_C,_C,_C,_C,_C,_C, /* 0-7 */ +_C,_C|_S,_C|_S,_C|_S,_C|_S,_C|_S,_C,_C, /* 8-15 */ +_C,_C,_C,_C,_C,_C,_C,_C, /* 16-23 */ +_C,_C,_C,_C,_C,_C,_C,_C, /* 24-31 */ +_S|_SP,_P,_P,_P,_P,_P,_P,_P, /* 32-39 */ +_P,_P,_P,_P,_P,_P,_P,_P, /* 40-47 */ +_D,_D,_D,_D,_D,_D,_D,_D, /* 48-55 */ +_D,_D,_P,_P,_P,_P,_P,_P, /* 56-63 */ +_P,_U|_X,_U|_X,_U|_X,_U|_X,_U|_X,_U|_X,_U, /* 64-71 */ +_U,_U,_U,_U,_U,_U,_U,_U, /* 72-79 */ +_U,_U,_U,_U,_U,_U,_U,_U, /* 80-87 */ +_U,_U,_U,_P,_P,_P,_P,_P, /* 88-95 */ +_P,_L|_X,_L|_X,_L|_X,_L|_X,_L|_X,_L|_X,_L, /* 96-103 */ +_L,_L,_L,_L,_L,_L,_L,_L, /* 104-111 */ +_L,_L,_L,_L,_L,_L,_L,_L, /* 112-119 */ +_L,_L,_L,_P,_P,_P,_P,_C, /* 120-127 */ +0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, /* 128-143 */ +0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0, /* 144-159 */ +_S|_SP,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P, /* 160-175 */ +_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P,_P, /* 176-191 */ +_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U,_U, /* 192-207 */ +_U,_U,_U,_U,_U,_U,_U,_P,_U,_U,_U,_U,_U,_U,_U,_L, /* 208-223 */ +_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L,_L, /* 224-239 */ +_L,_L,_L,_L,_L,_L,_L,_P,_L,_L,_L,_L,_L,_L,_L,_L}; /* 240-255 */ + int snd_pcm_format_width(int format) { @@ -46,6 +75,9 @@ int snd_pcm_format_width(int format) int snd_hda_create_hwdep(struct hda_codec *codec) { +#ifdef HAVE_USER_MUTEX + mutex_init(&codec->user_mutex); +#endif return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/2] A couple of pending HD-audio patches
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, these are yet another couple of HD-audio patches, one for merging to 3.8, and another an enhancement for 3.9. Takashi

1 2

[alsa-devel] [PATCH 00/12] Use standard fixup table for STAC/IDT codecs
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, this is a series of patches to convert the STAC/IDT codec parser to use the standard fixup table. Obviously it's a step forward for the migration of patch_sigmatel.c to the generic parser. Takashi

1 13

[alsa-devel] [PATCH RFC 0/2] Yet more patches for HD-audio generic parser
by Takashi Iwai 15 Jan '13

15 Jan '13

Hi, these two patches have slipped from the previous patch series. Takashi

1 2

[alsa-devel] sound/pci/hda/patch_sigmatel.c:2219:24: warning: unused variable 'spec'
by Fengguang Wu 15 Jan '13

15 Jan '13

Hi Takashi, FYI, there are new compile warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 629965377895ebe6a546184911d588fd29596dc8 commit: 8018ee4e722f6c408d475262b54dc8a73fdd349f ALSA: hda - Use standard fixup table for IDT92HD71Bxx date: 2 hours ago config: make ARCH=i386 defconfig All warnings: sound/pci/hda/patch_sigmatel.c: In function 'stac92hd71bxx_fixup_hp': >> sound/pci/hda/patch_sigmatel.c:2219:24: warning: unused variable 'spec' [-Wunused-variable] vim +/spec +2219 sound/pci/hda/patch_sigmatel.c 8018ee4e Takashi Iwai 2013-01-14 2216 static void stac92hd71bxx_fixup_hp(struct hda_codec *codec, 8018ee4e Takashi Iwai 2013-01-14 2217 const struct hda_fixup *fix, int action) 8018ee4e Takashi Iwai 2013-01-14 2218 { 8018ee4e Takashi Iwai 2013-01-14 @2219 struct sigmatel_spec *spec = codec->spec; 8018ee4e Takashi Iwai 2013-01-14 2220 8018ee4e Takashi Iwai 2013-01-14 2221 if (action != HDA_FIXUP_ACT_PRE_PROBE) 8018ee4e Takashi Iwai 2013-01-14 2222 return; 8018ee4e Takashi Iwai 2013-01-14 2223 8018ee4e Takashi Iwai 2013-01-14 2224 if (hp_blike_system(codec->subsystem_id)) { 8018ee4e Takashi Iwai 2013-01-14 2225 unsigned int pin_cfg = snd_hda_codec_get_pincfg(codec, 0x0f); 8018ee4e Takashi Iwai 2013-01-14 2226 if (get_defcfg_device(pin_cfg) == AC_JACK_LINE_OUT || 8018ee4e Takashi Iwai 2013-01-14 2227 get_defcfg_device(pin_cfg) == AC_JACK_SPEAKER || --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 1

[alsa-devel] sound/pci/hda/patch_sigmatel.c:2260:26: sparse: obsolete array initializer, use C99 syntax
by Fengguang Wu 15 Jan '13

15 Jan '13

Hi Takashi, FYI, there are new sparse warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 629965377895ebe6a546184911d588fd29596dc8 commit: 8018ee4e722f6c408d475262b54dc8a73fdd349f ALSA: hda - Use standard fixup table for IDT92HD71Bxx date: 2 hours ago >> sound/pci/hda/patch_sigmatel.c:2260:26: sparse: obsolete array initializer, use C99 syntax sound/pci/hda/patch_sigmatel.c:2266:26: sparse: obsolete array initializer, use C99 syntax vim +2260 sound/pci/hda/patch_sigmatel.c 8018ee4e Takashi Iwai 2013-01-14 2249 static const struct hda_fixup stac92hd71bxx_fixups[] = { 8018ee4e Takashi Iwai 2013-01-14 2250 [STAC_92HD71BXX_REF] = { 8018ee4e Takashi Iwai 2013-01-14 2251 .type = HDA_FIXUP_FUNC, 8018ee4e Takashi Iwai 2013-01-14 2252 .v.func = stac92hd71bxx_fixup_ref, 8018ee4e Takashi Iwai 2013-01-14 2253 }, 8018ee4e Takashi Iwai 2013-01-14 2254 [STAC_DELL_M4_1] = { 8018ee4e Takashi Iwai 2013-01-14 2255 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2256 .v.pins = dell_m4_1_pin_configs, 8018ee4e Takashi Iwai 2013-01-14 2257 .chained = true, 8018ee4e Takashi Iwai 2013-01-14 2258 .chain_id = STAC_92HD71BXX_NO_SMUX, 8018ee4e Takashi Iwai 2013-01-14 2259 }, 8018ee4e Takashi Iwai 2013-01-14 @2260 [STAC_DELL_M4_2] { 8018ee4e Takashi Iwai 2013-01-14 2261 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2262 .v.pins = dell_m4_2_pin_configs, 8018ee4e Takashi Iwai 2013-01-14 2263 .chained = true, 8018ee4e Takashi Iwai 2013-01-14 2264 .chain_id = STAC_92HD71BXX_NO_DMIC, 8018ee4e Takashi Iwai 2013-01-14 2265 }, 8018ee4e Takashi Iwai 2013-01-14 2266 [STAC_DELL_M4_3] { 8018ee4e Takashi Iwai 2013-01-14 2267 .type = HDA_FIXUP_PINS, 8018ee4e Takashi Iwai 2013-01-14 2268 .v.pins = dell_m4_3_pin_configs, --- 0-DAY kernel build testing backend Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 1

[alsa-devel] [PATCH 1/4] ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes
by Adrian Knoth 15 Jan '13

15 Jan '13

The current iobox detection code reportedly fails for various users, so simply do what the Win32 driver does instead. Patch originally by Karl Grill <kgrill(a)chello.at> and then modified to comply with kernel coding guidelines + current HEAD. Signed-off-by: Karl Grill <kgrill(a)chello.at> Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4fae81f..bcad9ba 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -154,10 +154,11 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_BIGENDIAN_MODE 0x200 #define HDSP_RD_MULTIPLE 0x400 #define HDSP_9652_ENABLE_MIXER 0x800 +#define HDSP_DUNNO 0x1000 #define HDSP_TDO 0x10000000 -#define HDSP_S_PROGRAM (HDSP_PROGRAM|HDSP_CONFIG_MODE_0) -#define HDSP_S_LOAD (HDSP_PROGRAM|HDSP_CONFIG_MODE_1) +#define HDSP_S_PROGRAM (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_0) +#define HDSP_S_LOAD (HDSP_DUNNO|HDSP_PROGRAM|HDSP_CONFIG_MODE_1) /* Control Register bits */ @@ -671,13 +672,23 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { + int i; + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; - if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { - snd_printk("Hammerfall-DSP: no IO box connected!\n"); - hdsp->state &= ~HDSP_FirmwareLoaded; - return -EIO; + for (i = 0; i < 500; i++) { + if (0 == (hdsp_read(hdsp, HDSP_statusRegister) & + HDSP_ConfigError)) { + if (i) { + snd_printk("Hammerfall-DSP: IO box found after %d ms\n", + i); + } + return 0; + } + msleep(20); } - return 0; + snd_printk("Hammerfall-DSP: no IO box connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, @@ -728,6 +739,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); return -EIO; } @@ -737,17 +749,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { hdsp_write(hdsp, HDSP_fifoData, cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout during firmware loading\n"); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); return -EIO; } } - ssleep(3); - - if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { - snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); - return -EIO; - } + hdsp_fifo_wait(hdsp, 3, HDSP_LONG_WAIT); + hdsp_write(hdsp, HDSP_control2Reg, 0x800); + ssleep(3); #ifdef SNDRV_BIG_ENDIAN hdsp->control2_register = HDSP_BIGENDIAN_MODE; #else @@ -773,24 +783,51 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); - hdsp_write (hdsp, HDSP_fifoData, 0); - if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) - return -EIO; + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); - hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + } + + hdsp_write(hdsp, HDSP_control2Reg, 0x820); hdsp_write (hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { - hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); - hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); - if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) - hdsp->io_type = RPM; - else - hdsp->io_type = Multiface; - } else { + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) == 0) { hdsp->io_type = Digiface; + snd_printk("Hammerfall-DSP: Digiface found\n"); + return 0; } + + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) == 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + + hdsp_write(hdsp, HDSP_control2Reg, 0x900); + hdsp_write(hdsp, HDSP_control2Reg, 0x10a0); + hdsp_write(hdsp, HDSP_fifoData, 0); + if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT) < 0) { + hdsp->io_type = Multiface; + snd_printk("Hammerfall-DSP: Multiface found\n"); + return 0; + } + + hdsp->io_type = RPM; + snd_printk("Hammerfall-DSP: RPM found\n"); + return 0; } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version2) -- 1.7.10.4

2 6

[alsa-devel] ASoC fixes for atmel
by Mark Brown 15 Jan '13

15 Jan '13

The following changes since commit 9931faca02c604c22335f5a935a501bb2ace6e20: Linux 3.8-rc3 (2013-01-09 18:59:55 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-atmel-pinctrl for you to fetch changes up to ea03c81521bde526570e1dec96eaa21fe5ac84a2: ASoC: atmel-ssc: add pinctrl selection to driver (2013-01-12 00:00:33 +0000) ---------------------------------------------------------------- ASoC: atmel: Fixes for pinctrl Due to a series of problems with the handling of Atmel, a combination of making changes that make other branches instantly buggy and a general failure to deal with the resulting issues effectively, v3.8 Atmel audio currently won't work at all for DT boards without adding pinctrl definitions and a request for those. ---------------------------------------------------------------- Bo Shen (2): ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: hda - mute LEDs on HP: also check PCI SSID
by David Henningsson 14 Jan '13

14 Jan '13

If the codec subvendor is not set to HP, maybe the PCI SSID is set to HP. If so, there might still be a valid mute LED string in BIOS/UEFI. Cc: stable(a)vger.kernel.org Buglink: https://bugs.launchpad.net/bugs/1099372 Reported-and-tested-by: Tammy Yang <tammy.yang(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a86547c..40c74a2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4948,7 +4948,8 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) &spec->gpio_led_polarity); return 1; } - if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { + if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP || + codec->bus->pci->subsystem_vendor == PCI_VENDOR_ID_HP) { while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (sscanf(dev->name, "HP_Mute_LED_%d_%x", -- 1.7.9.5

2 3

[alsa-devel] [RFC PATCH] ALSA: usb - fix race in creation of M-Audio Fast track pro driver
by David Henningsson 14 Jan '13

14 Jan '13

A patch in the 3.2 kernel caused regression with hotplugging the M-Audio Fast track pro, or sound after suspend. I don't have the device so I haven't done a full analysis, but it seems userspace (both udev and pulseaudio) got confused when a card was created, immediately destroyed, and then created again. However, at least one person in the bug report (martin djfun) reports that this patch resolves the issue for him. It also leaves a message in the log: "snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is a bit misleading. It is better than non-working audio, but maybe there's a more elegant solution? BugLink: https://bugs.launchpad.net/bugs/1095315 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/usb/quirks.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) ...oh, and if you end up taking this patch as it is, you should probably add cc to stable (3.2+). diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acc12f0..e71fc8b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -387,11 +387,13 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) * rules */ err = usb_driver_set_configuration(dev, 2); - if (err < 0) { + if (err < 0) snd_printdd("error usb_driver_set_configuration: %d\n", err); - return -ENODEV; - } + /* Always return an error, so that we stop creating a device + that will just be destroyed and recreated with a new + configuration */ + return -ENODEV; } else snd_printk(KERN_INFO "usb-audio: Fast Track Pro config OK\n"); -- 1.7.9.5

3 4

[alsa-devel] [PATCH 0/3] M-Audio C400 corrections
by Eldad Zack 14 Jan '13

14 Jan '13

Hi Takashi, The following are specific fixes for the M-Audio FT C400. * Use the clock selector instead of the internal clock and add names for the clocks instead of Unit 129 and Unit 130. * Fix an issue where the latency would change every time the stream is started. Uses skip_packets. I think they could go in current mainline, since it is only for this device, but I also checked that they apply cleanly against for-next. Cheers, Eldad Eldad Zack (3): ALSA: usb-audio: correct M-Audio C400 clock source quirk ALSA: usb-audio: M-Audio FT C400 skip packet quirk ALSA: usb-audio: selector map for M-Audio FT C400 sound/usb/mixer_maps.c | 13 +++++++++++++ sound/usb/quirks-table.h | 4 ++-- sound/usb/quirks.c | 11 +++++++++++ 3 files changed, 26 insertions(+), 2 deletions(-) -- 1.7.12.4

2 4

Re: [alsa-devel] [RFC PATCH 4/4] alsa/soc: add hdmi audio codec based on cdf
by Sachin Kamat 14 Jan '13

14 Jan '13

+CC: ALSA mailing list, Mark Brown On 13 January 2013 18:22, Rahul Sharma <rahul.sharma(a)samsung.com> wrote: > This patch registers hdmi-audio codec to the ALSA framework. This is the second > client to the hdmi panel. Once notified by the CDF Core it proceeds towards > audio setting and audio control. It also subscribes for hpd notification to > implement hpd related audio requirements. > > Signed-off-by: Rahul Sharma <rahul.sharma(a)samsung.com> > --- > sound/soc/codecs/Kconfig | 4 + > sound/soc/codecs/Makefile | 2 + > sound/soc/codecs/exynos_hdmi_audio.c | 307 +++++++++++++++++++++++++++++++++++ > 3 files changed, 313 insertions(+) > create mode 100644 sound/soc/codecs/exynos_hdmi_audio.c > > diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig > index b92759a..93f3f6b 100644 > --- a/sound/soc/codecs/Kconfig > +++ b/sound/soc/codecs/Kconfig > @@ -496,3 +496,7 @@ config SND_SOC_ML26124 > > config SND_SOC_TPA6130A2 > tristate > + > +config SND_SOC_EXYNOS_HDMI_AUDIO > + tristate > + default y Do you want to enable this by default? Shouldn't this be depending on HDMI support? > diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile > index 9bd4d95..bfe93e6 100644 > --- a/sound/soc/codecs/Makefile > +++ b/sound/soc/codecs/Makefile > @@ -112,6 +112,7 @@ snd-soc-wm9705-objs := wm9705.o > snd-soc-wm9712-objs := wm9712.o > snd-soc-wm9713-objs := wm9713.o > snd-soc-wm-hubs-objs := wm_hubs.o > +snd-soc-exynos-hdmi-audio-objs := exynos_hdmi_audio.o > > # Amp > snd-soc-max9877-objs := max9877.o > @@ -230,6 +231,7 @@ obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o > obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o > obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o > obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o > +obj-$(CONFIG_SND_SOC_EXYNOS_HDMI_AUDIO) += snd-soc-exynos-hdmi-audio.o > > # Amp > obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o > diff --git a/sound/soc/codecs/exynos_hdmi_audio.c b/sound/soc/codecs/exynos_hdmi_audio.c > new file mode 100644 > index 0000000..50e8564 > --- /dev/null > +++ b/sound/soc/codecs/exynos_hdmi_audio.c > @@ -0,0 +1,307 @@ > +/* > + * ALSA SoC codec driver for HDMI audio on Samsung Exynos processors. > + * Copyright (C) 2012 Samsung corp. Copyright (c) 2012 (-13?) Samsung Electronics Co., Ltd. > + * Author: Rahul Sharma <rahul.sharma(a)samsung.com> > + * > + * This program is free software; you can redistribute it and/or > + * modify it under the terms of the GNU General Public License > + * version 2 as published by the Free Software Foundation. > + * > + * This program is distributed in the hope that it will be useful, but > + * WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * General Public License for more details. > + * > + */ > +#include <linux/module.h> > +#include <linux/delay.h> > +#include <sound/soc.h> > +#include <video/display.h> > +#include <video/exynos_hdmi.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <linux/of_platform.h> > +#include <linux/platform_device.h> > + > +#undef dev_info > + > +#define dev_info(dev, format, arg...) \ > + dev_printk(KERN_CRIT, dev, format, ##arg) You may directly use dev_crit instead of dev_printk. > + > +static struct snd_soc_codec_driver hdmi_codec; > + > +/* platform device pointer for eynos hdmi audio device. */ > +static struct platform_device *exynos_hdmi_audio_pdev; > + > +struct hdmi_audio_context { > + struct platform_device *pdev; > + atomic_t plugged; > + struct display_entity_audio_params audio_params; > + struct display_entity *entity; > + struct display_entity_notifier notf; > + struct display_event_subscriber subscriber; > +}; > + > +static int exynos_hdmi_audio_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, > + struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct hdmi_audio_context *ctx = snd_soc_codec_get_drvdata(codec); > + int ret; > + > + dev_info(codec->dev, "[%d] %s\n", __LINE__, __func__); How about making this a debug message as it does not convey anything useful? > + > + ctx->audio_params.type = DISPLAY_ENTITY_AUDIO_I2S; > + > + switch (params_channels(params)) { > + case 6: > + case 4: > + case 2: > + case 1: > + ctx->audio_params.channels = params_channels(params); > + break; > + default: > + dev_err(codec->dev, "%d channels not supported\n", > + params_channels(params)); > + return -EINVAL; > + } > + > + switch (params_format(params)) { > + case SNDRV_PCM_FORMAT_S8: > + ctx->audio_params.bits_per_sample = 8; > + break; > + case SNDRV_PCM_FORMAT_S16_LE: > + ctx->audio_params.bits_per_sample = 12; > + break; > + case SNDRV_PCM_FORMAT_S24_LE: > + ctx->audio_params.bits_per_sample = 16; > + break; > + default: > + dev_err(codec->dev, "Format(%d) not supported\n", > + params_format(params)); > + return -EINVAL; > + } > + > + switch (params_rate(params)) { > + case 32000: > + case 44100: > + case 88200: > + case 176400: > + case 48000: > + case 96000: > + case 192000: > + ctx->audio_params.sf = params_rate(params); > + break; > + default: > + dev_err(codec->dev, "%d Rate supported\n", > + params_rate(params)); > + return -EINVAL; > + } > + > + /* checking here to cache audioparms for hpd plug handling */ > + if (!atomic_read(&ctx->plugged)) > + return -EINVAL; > + > + ret = > + display_entity_hdmi_init_audio(ctx->entity, &ctx->audio_params); > + return ret; > +} > + > +static int exynos_hdmi_audio_trigger(struct snd_pcm_substream *substream, > + int cmd, struct snd_soc_dai *dai) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct hdmi_audio_context *ctx = snd_soc_codec_get_drvdata(codec); > + int ret; > + > + dev_info(codec->dev, "[%d] %s\n", __LINE__, __func__); ditto > + > + /* checking here to cache audioparms for hpd plug handling */ > + if (!atomic_read(&ctx->plugged)) > + return -EINVAL; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + ret = display_entity_hdmi_set_audiostate(ctx->entity, > + DISPLAY_ENTITY_AUDIOSTATE_ON); > + if (ret) { > + dev_err(codec->dev, "audio enable failed.\n"); > + return -EINVAL; > + } > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + ret = display_entity_hdmi_set_audiostate(ctx->entity, > + DISPLAY_ENTITY_AUDIOSTATE_OFF); > + break; > + default: > + ret = -EINVAL; > + break; > + } > + > + return ret; > +} > + > +static const struct snd_soc_dai_ops exynos_hdmi_audio_dai_ops = { > + .hw_params = exynos_hdmi_audio_hw_params, > + .trigger = exynos_hdmi_audio_trigger, > +}; > + > +static struct snd_soc_dai_driver hdmi_codec_dai = { > + .name = "exynos-hdmi-audio", > + .playback = { > + .channels_min = 2, > + .channels_max = 8, > + .rates = SNDRV_PCM_RATE_32000 | > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | > + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | > + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE | > + SNDRV_PCM_FMTBIT_S24_LE, > + }, > + .ops = &exynos_hdmi_audio_dai_ops, > +}; > + > +void hdmi_audio_event_notify(struct display_entity *entity, > + enum display_entity_event_type type, > + unsigned int value, void *context) > +{ > + struct hdmi_audio_context *ctx = (struct hdmi_audio_context *)context; > + > + if (type == DISPLAY_ENTITY_HDMI_HOTPLUG) { > + dev_info(&ctx->pdev->dev, "[%d][%s] hpd(%d)\n", __LINE__, > + __func__, value); > + atomic_set(&ctx->plugged, !!value); > + } > +} > + > +int exynos_hdmi_audio_notification(struct display_entity_notifier *notf, > + struct display_entity *entity, int status) > +{ > + struct hdmi_audio_context *ctx = container_of(notf, > + struct hdmi_audio_context, notf); > + struct exynos_hdmi_control_ops *exynos_ops = > + (struct exynos_hdmi_control_ops *)entity->private; > + int hpd; > + > + if (status != DISPLAY_ENTITY_NOTIFIER_CONNECT && entity) > + return -EINVAL; > + > + dev_info(&ctx->pdev->dev, "[%d][%s]\n", __LINE__, __func__); > + > + ctx->entity = entity; > + > + ctx->subscriber.context = ctx; > + ctx->subscriber.notify = hdmi_audio_event_notify; > + > + display_entity_subscribe_event(entity, &ctx->subscriber); > + > + exynos_ops->get_hpdstate(entity, &hpd); > + atomic_set(&ctx->plugged, !!hpd); > + > + return 0; > +} > + > +static __devinit int hdmi_codec_probe(struct platform_device *pdev) __devinit is not necessary. +{ > + int ret; > + struct hdmi_audio_context *ctx; > + struct device_node *dev_node; > + struct platform_device *disp_pdev; > + > + dev_info(&pdev->dev, "[%d][%s]\n", __LINE__, __func__); > + > + ret = snd_soc_register_codec(&pdev->dev, &hdmi_codec, > + &hdmi_codec_dai, 1); > + > + if (ret) { > + dev_err(&pdev->dev, "register_codec failed (%d)\n", ret); > + return ret; > + } > + > + ctx = devm_kzalloc(&pdev->dev, sizeof(struct hdmi_audio_context), > + GFP_KERNEL); > + if (ctx == NULL) > + return -ENOMEM; > + > + ctx->pdev = pdev; > + atomic_set(&ctx->plugged, 0); > + platform_set_drvdata(pdev, ctx); > + > + dev_node = of_find_compatible_node(NULL, NULL, > + "samsung,exynos5-hdmi"); > + if (!dev_node) { > + dev_err(&pdev->dev, "[%d][%s] dt node not found.\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + > + disp_pdev = of_find_device_by_node(dev_node); > + if (!disp_pdev) { > + dev_err(&pdev->dev, "[ERROR][%d][%s] No pdev\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + > + ctx->notf.dev = &disp_pdev->dev; > + ctx->notf.notify = exynos_hdmi_audio_notification; > + > + ret = display_entity_register_notifier(&ctx->notf); > + if (ret) { > + dev_err(&pdev->dev, "[%d][%s] entity registe failed.\n", > + __LINE__, __func__); > + return -EINVAL; > + } > + return ret; > +} > + > +static __devexit int hdmi_codec_remove(struct platform_device *pdev) __devexit is not necessary. > +{ > + dev_info(&pdev->dev, " %s:%s:%d", __FILE__, __func__, __LINE__); > + mdelay(1000); > + > + snd_soc_unregister_codec(&pdev->dev); > + return 0; > +} > + > +static struct platform_driver hdmi_codec_driver = { > + .driver = { > + .name = "exynos-hdmi-audio-codec", > + .owner = THIS_MODULE, > + }, > + > + .probe = hdmi_codec_probe, > + .remove = __devexit_p(hdmi_codec_remove), > +}; > + > +static int __init hdmi_codec_init(void) > +{ > + int ret; > + > + ret = platform_driver_register(&hdmi_codec_driver); > + if (ret < 0) > + return -EINVAL; > + > + exynos_hdmi_audio_pdev = platform_device_register_simple > + ("exynos-hdmi-audio-codec", -1, NULL, 0); > + if (IS_ERR_OR_NULL(exynos_hdmi_audio_pdev)) { > + ret = PTR_ERR(exynos_hdmi_audio_pdev); > + platform_driver_unregister(&hdmi_codec_driver); > + return ret; > + } > + > + return 0; > +} > +static void __exit hdmi_codec_exit(void) > +{ > + platform_driver_unregister(&hdmi_codec_driver); > + platform_device_unregister(exynos_hdmi_audio_pdev); > +} > + > +module_init(hdmi_codec_init); > +module_exit(hdmi_codec_exit); > + > +MODULE_AUTHOR("Rahul Sharma <rahul.sharma(a)samsung.com>"); > +MODULE_DESCRIPTION("ASoC EXYNOS HDMI codec driver"); > +MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:" DRV_NAME); > -- > 1.8.0 > > -- > To unsubscribe from this list: send the line "unsubscribe linux-media" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html -- With warm regards, Sachin

2 1

[alsa-devel] [PATCH 0/2] ASoC: fsi: add DT support
by Kuninori Morimoto 14 Jan '13

14 Jan '13

Hi Mark These patches adds device tree support on FSI driver These are based on mark/topic/fsi branch Kuninori Morimoto (2): ASoC: fsi: don't use platform info pointer on probe() ASoC: fsi: add device tree support .../devicetree/bindings/sound/renesas,fsi.txt | 23 +++++ sound/soc/sh/fsi.c | 98 ++++++++++++++++---- 2 files changed, 104 insertions(+), 17 deletions(-) Best regards --- Kuninori Morimoto

2 7

Re: [alsa-devel] [PATCH v2 3/3] ASoC: tegra: add tegra machine driver using wm9712 codec
by Mark Brown 14 Jan '13

14 Jan '13

On Tue, Jan 08, 2013 at 11:52:33PM +0100, Lucas Stach wrote: > Am Dienstag, den 08.01.2013, 15:17 -0700 schrieb Stephen Warren: > > Hmmm. I thought that could be auto-instantiated based on probing the > > AC'97 bus for whatever was there? > AC97 is in principle probeable by reading the first regs of the codec, > but all the infrastructure to do proper hotplugging on the AC97 bus is > missing in the kernel. As I didn't want to be the one to add all this > infrastructure to a dying audio platform, I opted to do the same thing > as all the other platform using AC97 and instantiate the codec from the > machine driver. That's about the size of it, yes. We should write the device tree as though we'd got better infrastructure here since this isn't something that ought to be required but right now this is how you do AC'97 in ASoC. It'd be really nice if someone had the time to work on the AC'97 infrastructure here but I'd not hold my breath given that it's rarely used in new designs.

1 0

[alsa-devel] [PATCH 1/8] regmap: Add support for 24 bit wide register addresses
by Lars-Peter Clausen 13 Jan '13

13 Jan '13

Since regmap already has support for formatting 24 bit wide values, so adding support for 24 bit wide registers is pretty much straight forward. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- drivers/base/regmap/regmap.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 42d5cb0..26af93a 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -500,6 +500,12 @@ struct regmap *regmap_init(struct device *dev, } break; + case 24: + if (reg_endian != REGMAP_ENDIAN_BIG) + goto err_map; + map->format.format_reg = regmap_format_24; + break; + case 32: switch (reg_endian) { case REGMAP_ENDIAN_BIG: -- 1.8.0

2 17

[alsa-devel] [PATCH] ASoC: simple-card: add asoc_simple_dai for initializing
by Kuninori Morimoto 13 Jan '13

13 Jan '13

Current simple-card driver calls asoc_simple_card_dai_init() if platform had a asoc_simple_card_dai_init pointer. And, this initialization function works only when platform has an applicable initial value for each dai settings. And basically, almost all sound card requires certain initialization. This means that almost all platform has initialization settings, and driver do nothing if it doesn't have settings. And additionally, current simple-card supports sysclk settings but it was only for codec. In order to abolish deviation between cpu and codec, and in order to simplify processing, this patch adds asoc_simple_dai, and removed pointless struct asoc_simple_dai_init_info which was trigger of calling asoc_simple_card_dai_init(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark This patch is based on both mark/topic/simple-card and mark/topic/fsi. I used "git merge". Please let me know if I should rebase it. >> Simon Please give this patch your Acked-by, since it modifies mach-shmobile. arch/arm/mach-shmobile/board-ap4evb.c | 35 +++++++------- arch/arm/mach-shmobile/board-armadillo800eva.c | 34 +++++++------- arch/arm/mach-shmobile/board-kzm9g.c | 20 ++++---- arch/arm/mach-shmobile/board-mackerel.c | 35 +++++++------- arch/sh/boards/mach-ecovec24/setup.c | 19 ++++---- arch/sh/boards/mach-se/7724/setup.c | 21 ++++----- include/sound/simple_card.h | 12 ++--- sound/soc/generic/simple-card.c | 58 ++++++++++++------------ 8 files changed, 116 insertions(+), 118 deletions(-) diff --git a/arch/arm/mach-shmobile/board-ap4evb.c b/arch/arm/mach-shmobile/board-ap4evb.c index 4c97903..08294fa 100644 --- a/arch/arm/mach-shmobile/board-ap4evb.c +++ b/arch/arm/mach-shmobile/board-ap4evb.c @@ -686,21 +686,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { @@ -809,19 +809,18 @@ static struct platform_device lcdc1_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 042a21e..22a1cc5 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -807,21 +807,21 @@ static struct platform_device fsi_device = { }; /* FSI-WM8978 */ -static struct asoc_simple_dai_init_info fsi_wm8978_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 12288000, -}; - static struct asoc_simple_card_info fsi_wm8978_info = { .name = "wm8978", .card = "FSI2A-WM8978", - .cpu_dai = "fsia-dai", .codec = "wm8978.0-001a", .platform = "sh_fsi2", - .codec_dai = "wm8978-hifi", - .init = &fsi_wm8978_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "wm8978-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_NB_NF, + .sysclk = 12288000, + }, }; static struct platform_device fsi_wm8978_device = { @@ -833,18 +833,18 @@ static struct platform_device fsi_wm8978_device = { }; /* FSI-HDMI */ -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { diff --git a/arch/arm/mach-shmobile/board-kzm9g.c b/arch/arm/mach-shmobile/board-kzm9g.c index 9ca7ce5..f170aed 100644 --- a/arch/arm/mach-shmobile/board-kzm9g.c +++ b/arch/arm/mach-shmobile/board-kzm9g.c @@ -526,21 +526,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4648_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4648_info = { .name = "AK4648", .card = "FSI2A-AK4648", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4648_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4648_device = { diff --git a/arch/arm/mach-shmobile/board-mackerel.c b/arch/arm/mach-shmobile/board-mackerel.c index b5d210b..3fd716d 100644 --- a/arch/arm/mach-shmobile/board-mackerel.c +++ b/arch/arm/mach-shmobile/board-mackerel.c @@ -502,19 +502,18 @@ static struct platform_device hdmi_lcdc_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_hdmi_init_info = { - .cpu_daifmt = SND_SOC_DAIFMT_CBM_CFM | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi2_hdmi_info = { .name = "HDMI", .card = "FSI2B-HDMI", - .cpu_dai = "fsib-dai", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", - .codec_dai = "sh_mobile_hdmi-hifi", - .init = &fsi2_hdmi_init_info, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "sh_mobile_hdmi-hifi", + }, }; static struct platform_device fsi_hdmi_device = { @@ -893,21 +892,21 @@ static struct platform_device fsi_device = { }, }; -static struct asoc_simple_dai_init_info fsi2_ak4643_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi2_ak4643_info = { .name = "AK4643", .card = "FSI2A-AK4643", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4643_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4643_device = { diff --git a/arch/sh/boards/mach-ecovec24/setup.c b/arch/sh/boards/mach-ecovec24/setup.c index 8ebe4c7..065e9600 100644 --- a/arch/sh/boards/mach-ecovec24/setup.c +++ b/arch/sh/boards/mach-ecovec24/setup.c @@ -897,21 +897,20 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi_da7210_init_info = { - .fmt = SND_SOC_DAIFMT_I2S, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, -}; - static struct asoc_simple_card_info fsi_da7210_info = { .name = "DA7210", .card = "FSIB-DA7210", - .cpu_dai = "fsib-dai", .codec = "da7210.0-001a", .platform = "sh_fsi.0", - .codec_dai = "da7210-hifi", - .init = &fsi_da7210_init_info, + .daifmt = SND_SOC_DAIFMT_I2S, + .cpu_dai = { + .name = "fsib-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "da7210-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + }, }; static struct platform_device fsi_da7210_device = { diff --git a/arch/sh/boards/mach-se/7724/setup.c b/arch/sh/boards/mach-se/7724/setup.c index 975608f..4010e63 100644 --- a/arch/sh/boards/mach-se/7724/setup.c +++ b/arch/sh/boards/mach-se/7724/setup.c @@ -299,22 +299,21 @@ static struct platform_device fsi_device = { .resource = fsi_resources, }; -static struct asoc_simple_dai_init_info fsi2_ak4642_init_info = { - .fmt = SND_SOC_DAIFMT_LEFT_J, - .codec_daifmt = SND_SOC_DAIFMT_CBM_CFM, - .cpu_daifmt = SND_SOC_DAIFMT_CBS_CFS | - SND_SOC_DAIFMT_IB_NF, - .sysclk = 11289600, -}; - static struct asoc_simple_card_info fsi_ak4642_info = { .name = "AK4642", .card = "FSIA-AK4642", - .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", - .codec_dai = "ak4642-hifi", - .init = &fsi2_ak4642_init_info, + .daifmt = SND_SOC_DAIFMT_LEFT_J, + .cpu_dai = { + .name = "fsia-dai", + .fmt = SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF, + }, + .codec_dai = { + .name = "ak4642-hifi", + .fmt = SND_SOC_DAIFMT_CBM_CFM, + .sysclk = 11289600, + }, }; static struct platform_device fsi_ak4642_device = { diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 4b62b8d..6c74527 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -14,21 +14,21 @@ #include <sound/soc.h> -struct asoc_simple_dai_init_info { +struct asoc_simple_dai { + const char *name; unsigned int fmt; - unsigned int cpu_daifmt; - unsigned int codec_daifmt; unsigned int sysclk; }; struct asoc_simple_card_info { const char *name; const char *card; - const char *cpu_dai; const char *codec; const char *platform; - const char *codec_dai; - struct asoc_simple_dai_init_info *init; /* for snd_link.init */ + + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; /* used in simple-card.c */ struct snd_soc_dai_link snd_link; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bc050ec..6cf8355 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -16,33 +16,38 @@ #define asoc_simple_get_card_info(p) \ container_of(p->dai_link, struct asoc_simple_card_info, snd_link) +static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, + struct asoc_simple_dai *set, + unsigned int daifmt) +{ + int ret = 0; + + daifmt |= set->fmt; + + if (!ret && daifmt) + ret = snd_soc_dai_set_fmt(dai, daifmt); + + if (!ret && set->sysclk) + ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + + return ret; +} + static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *cinfo = asoc_simple_get_card_info(rtd); - struct asoc_simple_dai_init_info *iinfo = cinfo->init; + struct asoc_simple_card_info *info = asoc_simple_get_card_info(rtd); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int cpu_daifmt = iinfo->fmt | iinfo->cpu_daifmt; - unsigned int codec_daifmt = iinfo->fmt | iinfo->codec_daifmt; + unsigned int daifmt = info->daifmt; int ret; - if (codec_daifmt) { - ret = snd_soc_dai_set_fmt(codec, codec_daifmt); - if (ret < 0) - return ret; - } - - if (iinfo->sysclk) { - ret = snd_soc_dai_set_sysclk(codec, 0, iinfo->sysclk, 0); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + if (ret < 0) + return ret; - if (cpu_daifmt) { - ret = snd_soc_dai_set_fmt(cpu, cpu_daifmt); - if (ret < 0) - return ret; - } + ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + if (ret < 0) + return ret; return 0; } @@ -59,10 +64,10 @@ static int asoc_simple_card_probe(struct platform_device *pdev) if (!cinfo->name || !cinfo->card || - !cinfo->cpu_dai || !cinfo->codec || !cinfo->platform || - !cinfo->codec_dai) { + !cinfo->cpu_dai.name || + !cinfo->codec_dai.name) { dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } @@ -72,14 +77,11 @@ static int asoc_simple_card_probe(struct platform_device *pdev) */ cinfo->snd_link.name = cinfo->name; cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai; + cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; cinfo->snd_link.platform_name = cinfo->platform; cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai; - - /* enable snd_link.init if cinfo has settings */ - if (cinfo->init) - cinfo->snd_link.init = asoc_simple_card_dai_init; + cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; + cinfo->snd_link.init = asoc_simple_card_dai_init; /* * init snd_soc_card -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Remove suspend/resume soc driver operations
by Peter Ujfalusi 13 Jan '13

13 Jan '13

With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320dac33.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 782b0cd..4f35839 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1452,20 +1452,6 @@ static int dac33_soc_remove(struct snd_soc_codec *codec) return 0; } -static int dac33_soc_suspend(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int dac33_soc_resume(struct snd_soc_codec *codec) -{ - dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, @@ -1476,8 +1462,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .reg_cache_default = dac33_reg, .probe = dac33_soc_probe, .remove = dac33_soc_remove, - .suspend = dac33_soc_suspend, - .resume = dac33_soc_resume, .controls = dac33_snd_controls, .num_controls = ARRAY_SIZE(dac33_snd_controls), -- 1.8.1

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Provide explicit trace of coefficient writes
by Mark Brown 12 Jan '13

12 Jan '13

Helpful for debugging. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4a233b8..14af8e2 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -775,6 +775,9 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) return -ENOMEM; } + adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", + file, blocks, le32_to_cpu(blk->len), + reg); ret = regmap_raw_write(regmap, reg, blk->data, le32_to_cpu(blk->len)); if (ret != 0) { -- 1.7.10.4

1 0

[alsa-devel] [PATCH 0/2 v2] at91/ssc: fixes on ASoC tree for 3.8
by Nicolas Ferre 12 Jan '13

12 Jan '13

Hi Mark, Olof, Arnd, This series goes on top of Linus' v3.8-rc3 and fixes an error that we have while compiling DTBs for AT91: ERROR (phandle_references): Reference to non-existent node or label "pinctrl_ssc0_tx" ERROR: Input tree has errors, aborting (use -f to force output) make[3]: *** [arch/arm/boot/dts/at91sam9g20ek.dtb] Error 2 I have included pending material that is dealing with SSC and pinctrl. The pinctrl part can be merged more easily now that the big pinctrl update for AT91 has been merged upstream. This material was designed to enter Mark's fixes queue, but as discussed with Olof, we can imagine merging everything through arm-soc or split the series (of 2 patches) and let them progress upstream separated (option that I do not like even if I know that the consequences are not so dramatic). So please, Olof, if you feel confortable with this series, tell us what you prefer and we will make our best to make this material go forward... Thanks for your help, best regards, v2: - remove the fix for typo in SSC status property: already merged upstream Bo Shen (2): ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-) -- 1.8.0

4 7

[alsa-devel] sound/pci/hda/hda_hwdep.c:487 parse_hints() warn: inconsistent returns mutex:&codec->user_mutex: locked (478) unlocked (447, 449, 452, 457, 487)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: bea06a147981e88465f566acd6c6e98b297f7712 ALSA: hda - Protect user-defined arrays via mutex date: 3 hours ago sound/pci/hda/hda_hwdep.c:487 parse_hints() warn: inconsistent returns mutex:&codec->user_mutex: locked (478) unlocked (447,449,452,457,487) vim +487 sound/pci/hda/hda_hwdep.c 43b62713 Takashi Iwai 2009-03-02 441 char *key, *val; 43b62713 Takashi Iwai 2009-03-02 442 struct hda_hint *hint; bea06a14 Takashi Iwai 2013-01-10 443 int err = 0; 1e1be432 Takashi Iwai 2008-07-30 444 e7d2860b André Goddard Rosa 2009-12-14 445 buf = skip_spaces(buf); 43b62713 Takashi Iwai 2009-03-02 446 if (!*buf || *buf == '#' || *buf == '\n') 4ea6fbc8 Takashi Iwai 2009-06-17 @447 return 0; 43b62713 Takashi Iwai 2009-03-02 448 if (*buf == '=') 43b62713 Takashi Iwai 2009-03-02 @449 return -EINVAL; 43b62713 Takashi Iwai 2009-03-02 450 key = kstrndup_noeol(buf, 1024); 43b62713 Takashi Iwai 2009-03-02 451 if (!key) 1e1be432 Takashi Iwai 2008-07-30 @452 return -ENOMEM; 43b62713 Takashi Iwai 2009-03-02 453 /* extract key and val */ 43b62713 Takashi Iwai 2009-03-02 454 val = strchr(key, '='); 43b62713 Takashi Iwai 2009-03-02 455 if (!val) { 43b62713 Takashi Iwai 2009-03-02 456 kfree(key); 43b62713 Takashi Iwai 2009-03-02 @457 return -EINVAL; 43b62713 Takashi Iwai 2009-03-02 458 } 43b62713 Takashi Iwai 2009-03-02 459 *val++ = 0; e7d2860b André Goddard Rosa 2009-12-14 460 val = skip_spaces(val); 43b62713 Takashi Iwai 2009-03-02 461 remove_trail_spaces(key); 43b62713 Takashi Iwai 2009-03-02 462 remove_trail_spaces(val); bea06a14 Takashi Iwai 2013-01-10 463 mutex_lock(&codec->user_mutex); 43b62713 Takashi Iwai 2009-03-02 464 hint = get_hint(codec, key); 43b62713 Takashi Iwai 2009-03-02 465 if (hint) { 43b62713 Takashi Iwai 2009-03-02 466 /* replace */ 43b62713 Takashi Iwai 2009-03-02 467 kfree(hint->key); 43b62713 Takashi Iwai 2009-03-02 468 hint->key = key; 43b62713 Takashi Iwai 2009-03-02 469 hint->val = val; bea06a14 Takashi Iwai 2013-01-10 470 goto unlock; 43b62713 Takashi Iwai 2009-03-02 471 } 43b62713 Takashi Iwai 2009-03-02 472 /* allocate a new hint entry */ 43b62713 Takashi Iwai 2009-03-02 473 if (codec->hints.used >= MAX_HINTS) 43b62713 Takashi Iwai 2009-03-02 474 hint = NULL; 43b62713 Takashi Iwai 2009-03-02 475 else 43b62713 Takashi Iwai 2009-03-02 476 hint = snd_array_new(&codec->hints); 1e1be432 Takashi Iwai 2008-07-30 477 if (!hint) { 1e1be432 Takashi Iwai 2008-07-30 @478 return -ENOMEM; bea06a14 Takashi Iwai 2013-01-10 479 goto unlock; 1e1be432 Takashi Iwai 2008-07-30 480 } 43b62713 Takashi Iwai 2009-03-02 481 hint->key = key; 43b62713 Takashi Iwai 2009-03-02 482 hint->val = val; bea06a14 Takashi Iwai 2013-01-10 483 unlock: bea06a14 Takashi Iwai 2013-01-10 484 mutex_unlock(&codec->user_mutex); bea06a14 Takashi Iwai 2013-01-10 485 if (err) bea06a14 Takashi Iwai 2013-01-10 486 kfree(key); bea06a14 Takashi Iwai 2013-01-10 @487 return err; 4ea6fbc8 Takashi Iwai 2009-06-17 488 } 4ea6fbc8 Takashi Iwai 2009-06-17 489 4ea6fbc8 Takashi Iwai 2009-06-17 490 static ssize_t hints_store(struct device *dev, --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

2 1

[alsa-devel] sound/pci/hda/hda_generic.c:1529 indep_hp_put() warn: inconsistent returns mutex:&spec->pcm_mutex: locked (1526) unlocked (1517, 1529)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: a5de6f4d021608c50ff5ff51d208b5030da92957 ALSA: hda - Implement independent HP control date: 33 hours ago sound/pci/hda/hda_generic.c:1529 indep_hp_put() warn: inconsistent returns mutex:&spec->pcm_mutex: locked (1526) unlocked (1517,1529) vim +1529 sound/pci/hda/hda_generic.c a5de6f4d Takashi Iwai 2012-12-21 1511 struct hda_gen_spec *spec = codec->spec; a5de6f4d Takashi Iwai 2012-12-21 1512 unsigned int select = ucontrol->value.enumerated.item[0]; a5de6f4d Takashi Iwai 2012-12-21 1513 a5de6f4d Takashi Iwai 2012-12-21 1514 mutex_lock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 1515 if (spec->active_streams) { a5de6f4d Takashi Iwai 2012-12-21 1516 mutex_unlock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 @1517 return -EBUSY; a5de6f4d Takashi Iwai 2012-12-21 1518 } a5de6f4d Takashi Iwai 2012-12-21 1519 a5de6f4d Takashi Iwai 2012-12-21 1520 if (spec->indep_hp_enabled != select) { a5de6f4d Takashi Iwai 2012-12-21 1521 spec->indep_hp_enabled = select; a5de6f4d Takashi Iwai 2012-12-21 1522 if (spec->indep_hp_enabled) a5de6f4d Takashi Iwai 2012-12-21 1523 spec->multiout.hp_out_nid[0] = 0; a5de6f4d Takashi Iwai 2012-12-21 1524 else a5de6f4d Takashi Iwai 2012-12-21 1525 spec->multiout.hp_out_nid[0] = spec->alt_dac_nid; a5de6f4d Takashi Iwai 2012-12-21 @1526 return 1; a5de6f4d Takashi Iwai 2012-12-21 1527 } a5de6f4d Takashi Iwai 2012-12-21 1528 mutex_unlock(&spec->pcm_mutex); a5de6f4d Takashi Iwai 2012-12-21 @1529 return 0; a5de6f4d Takashi Iwai 2012-12-21 1530 } a5de6f4d Takashi Iwai 2012-12-21 1531 a5de6f4d Takashi Iwai 2012-12-21 1532 static const struct snd_kcontrol_new indep_hp_ctl = { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] sound/pci/hda/hda_codec.c:3433 snd_hda_codec_write_cache() error: we previously assumed 'c' could be null (see line 3430)
by Fengguang Wu 12 Jan '13

12 Jan '13

Hi Takashi, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master head: 02e5aa7b459f69538d4ec88658d064816a7692d6 commit: 95836563e7f636fc57732f5e687ca2f170eb674f ALSA: hda - Introduce cache & flush cmd / amp writes date: 33 hours ago sound/pci/hda/hda_codec.c:3433 snd_hda_codec_write_cache() error: we previously assumed 'c' could be null (see line 3430) vim +/c +3433 sound/pci/hda/hda_codec.c aa2936f5 Takashi Iwai 2009-05-26 3424 /* parm may contain the verb stuff for get/set amp */ aa2936f5 Takashi Iwai 2009-05-26 3425 verb = verb | (parm >> 8); aa2936f5 Takashi Iwai 2009-05-26 3426 parm &= 0xff; aa2936f5 Takashi Iwai 2009-05-26 3427 key = build_cmd_cache_key(nid, verb); aa2936f5 Takashi Iwai 2009-05-26 3428 mutex_lock(&codec->bus->cmd_mutex); aa2936f5 Takashi Iwai 2009-05-26 3429 c = get_alloc_hash(&codec->cmd_cache, key); aa2936f5 Takashi Iwai 2009-05-26 @3430 if (c) aa2936f5 Takashi Iwai 2009-05-26 3431 c->val = parm; 95836563 Takashi Iwai 2012-12-13 3432 if (codec->cached_write) 95836563 Takashi Iwai 2012-12-13 @3433 c->dirty = 1; aa2936f5 Takashi Iwai 2009-05-26 3434 mutex_unlock(&codec->bus->cmd_mutex); aa2936f5 Takashi Iwai 2009-05-26 3435 return 0; b3ac5636 Takashi Iwai 2007-08-10 3436 } --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [RESEND PATCH] ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word
by Ricardo Neri 12 Jan '13

12 Jan '13

As the IEC-60958 channel status word is set by ANDing and ORing with the appropriate definitions, the word bytes need to be initialized to zero to avoid misconfiguration due to previous hw_params calls. Signed-off-by: Ricardo Neri <rneri(a)dextratech.com> --- sound/soc/omap/omap-hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index f59c69f..75ad378 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -110,6 +110,8 @@ static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream, /* * fill the IEC-60958 channel status word */ + /* initialize the word bytes */ + memset(iec->status, 0, sizeof(iec->status)); /* specify IEC-60958-3 (commercial use) */ iec->status[0] &= ~IEC958_AES0_PROFESSIONAL; -- 1.7.10.4

2 2

[alsa-devel] [patch] ASoC: pcm: delete some dead code
by Dan Carpenter 12 Jan '13

12 Jan '13

I've removed several unreachable returns. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6..1d6af4d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1729,20 +1729,16 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) /* startup must always be called for new BEs */ ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { + if (ret < 0) goto disconnect; - return ret; - } /* keep going if FE state is > open */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_OPEN) return 0; ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) { + if (ret < 0) goto close; - return ret; - } /* keep going if FE state is > hw_params */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_PARAMS) @@ -1750,10 +1746,8 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) { + if (ret < 0) goto hw_free; - return ret; - } /* run the stream event for each BE */ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_NOP);

2 1

[alsa-devel] [PATCH] ASoC: fsl: fix multiple definition of init_module
by Shawn Guo 12 Jan '13

12 Jan '13

With commit f2818d0 (ASoC: fsl: fix miscompilation of snd-soc-imx-pcm), we will see the following build error when building modules with CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig. CC [M] sound/soc/fsl/phycore-ac97.o LD [M] sound/soc/fsl/snd-soc-fsl-ssi.o LD [M] sound/soc/fsl/snd-soc-fsl-utils.o LD [M] sound/soc/fsl/snd-soc-imx-ssi.o LD [M] sound/soc/fsl/snd-soc-imx-audmux.o LD [M] sound/soc/fsl/snd-soc-imx-pcm.o sound/soc/fsl/imx-pcm-dma.o: In function `init_module': imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module' sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module': imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module' sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1 Instead of using bool for SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA to fix the original issue, we should completely remove SND_SOC_IMX_PCM and have imx-pcm.o statically linked with imx-pcm-fiq.o or imx-pcm-dma.o. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- Mark, To be clear, I think it's a v3.8 material. Shawn sound/soc/fsl/Kconfig | 9 ++------- sound/soc/fsl/Makefile | 5 ++++- sound/soc/fsl/imx-pcm.c | 3 --- 3 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3b98159..a210c8d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -108,18 +108,13 @@ if SND_IMX_SOC config SND_SOC_IMX_SSI tristate -config SND_SOC_IMX_PCM - tristate - config SND_SOC_IMX_PCM_FIQ - bool + tristate select FIQ - select SND_SOC_IMX_PCM config SND_SOC_IMX_PCM_DMA - bool + tristate select SND_SOC_DMAENGINE_PCM - select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index afd3479..ec14579 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -41,7 +41,10 @@ endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o -obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o +snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o +snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/fsl/imx-pcm.c b/sound/soc/fsl/imx-pcm.c index d5cd9ef..0c9f188 100644 --- a/sound/soc/fsl/imx-pcm.c +++ b/sound/soc/fsl/imx-pcm.c @@ -31,7 +31,6 @@ int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); return ret; } -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { @@ -80,7 +79,6 @@ int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) out: return ret; } -EXPORT_SYMBOL_GPL(imx_pcm_new); void imx_pcm_free(struct snd_pcm *pcm) { @@ -102,7 +100,6 @@ void imx_pcm_free(struct snd_pcm *pcm) buf->area = NULL; } } -EXPORT_SYMBOL_GPL(imx_pcm_free); MODULE_DESCRIPTION("Freescale i.MX PCM driver"); MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: twl6040: Updates for 3.9
by Peter Ujfalusi 12 Jan '13

12 Jan '13

Hi Mark, Liam, Changes since v1: - patch 2 for twl6040_resume() has been replaced with a simpler one Cover letter from v1: Small fixe from Misael for twl6040 codec driver and devm_* conversion, switch to system workqueue and dead code removal. Regards, Peter --- Misael Lopez Cruz (1): ASoC: twl6040: Convert PLUGINT to no-suspend irq Peter Ujfalusi (4): ASoC: twl6040: Only set the bias_level once in twl6040_resume() ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation sound/soc/codecs/twl6040.c | 62 +++++++--------------------------------------- 1 file changed, 9 insertions(+), 53 deletions(-) -- 1.8.1

2 6

[alsa-devel] [PATCH] ASoC: twl4030: Remove suspend/resume soc driver operations
by Peter Ujfalusi 12 Jan '13

12 Jan '13

With idle_bias_off these are no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl4030.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 63b280b..5611b7b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2267,18 +2267,6 @@ static struct snd_soc_dai_driver twl4030_dai[] = { }, }; -static int twl4030_soc_suspend(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int twl4030_soc_resume(struct snd_soc_codec *codec) -{ - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; -} - static int twl4030_soc_probe(struct snd_soc_codec *codec) { struct twl4030_priv *twl4030; @@ -2316,8 +2304,6 @@ static int twl4030_soc_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .probe = twl4030_soc_probe, .remove = twl4030_soc_remove, - .suspend = twl4030_soc_suspend, - .resume = twl4030_soc_resume, .read = twl4030_read_reg_cache, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, -- 1.8.1

2 1

Re: [alsa-devel] [PATCH v2 06/10] ASoC: tegra: remove auxdata
by Stephen Warren 11 Jan '13

11 Jan '13

On 01/11/2013 01:01 AM, Prashant Gaikwad wrote: > Configlink clock information is added to device tree. Get the clocks > using device node. Remove AUXDATA. Cc'ing in the ASoC maintainers as an FYI and for Acks; this patch needs to go through the Tegra tree due to dependencies. > > Signed-off-by: Prashant Gaikwad <pgaikwad(a)nvidia.com> > --- > sound/soc/tegra/tegra30_ahub.c | 14 ++------------ > 1 files changed, 2 insertions(+), 12 deletions(-) > > diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c > index bb31c41..2355630 100644 > --- a/sound/soc/tegra/tegra30_ahub.c > +++ b/sound/soc/tegra/tegra30_ahub.c > @@ -299,15 +299,6 @@ static const char * const configlink_clocks[] = { > "spdif_in", > }; > > -struct of_dev_auxdata ahub_auxdata[] = { > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080300, "tegra30-i2s.0", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080400, "tegra30-i2s.1", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080500, "tegra30-i2s.2", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080600, "tegra30-i2s.3", NULL), > - OF_DEV_AUXDATA("nvidia,tegra30-i2s", 0x70080700, "tegra30-i2s.4", NULL), > - {} > -}; > - > #define LAST_REG(name) \ > (TEGRA30_AHUB_##name + \ > (TEGRA30_AHUB_##name##_STRIDE * TEGRA30_AHUB_##name##_COUNT) - 4) > @@ -451,7 +442,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) > * Ensure that here. > */ > for (i = 0; i < ARRAY_SIZE(configlink_clocks); i++) { > - clk = clk_get_sys(NULL, configlink_clocks[i]); > + clk = clk_get(&pdev->dev, configlink_clocks[i]); > if (IS_ERR(clk)) { > dev_err(&pdev->dev, "Can't get clock %s\n", > configlink_clocks[i]); > @@ -569,8 +560,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) > goto err_pm_disable; > } > > - of_platform_populate(pdev->dev.of_node, NULL, ahub_auxdata, > - &pdev->dev); > + of_platform_populate(pdev->dev.of_node, NULL, NULL, &pdev->dev); > > return 0; > >

2 1

[alsa-devel] [PATCH 0/5] ASoC: twl6040: Updates for 3.9
by Peter Ujfalusi 11 Jan '13

11 Jan '13

Hi Mark, Liam, Small fixes from Misael for twl6040 codec driver and devm_* conversion, switch to system workqueue and dead code removal. Regards, Peter --- Misael Lopez Cruz (2): ASoC: twl6040: Convert PLUGINT to no-suspend irq ASoC: twl6040: Prevent extra power transitions during resume Peter Ujfalusi (3): ASoC: twl6040: Convert to use devm_* when possible ASoC: twl6040: Switch to use system workqueue for jack reporting ASoC: twl6040: Remove leftover code from hs/hf ramp implementation sound/soc/codecs/twl6040.c | 67 +++++++++------------------------------------- 1 file changed, 13 insertions(+), 54 deletions(-) -- 1.8.1

2 12

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 11 Jan '13

11 Jan '13

At Fri, 11 Jan 2013 08:47:25 +0100, Jens Axboe wrote: > > On 2013-01-10 21:19, Takashi Iwai wrote: > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH v2] ALSA: usb-audio: Fix NULL dereference by access to > > non-existing substream > > > > The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for > > audioformat mismatch] introduced the correction of parameters to be > > set for sync EP. But since the new code assumes that the sync EP is > > always paired with the data EP of another direction, it triggers Oops > > when a device only with a single direction is used. > > > > This patch adds a proper check of sync EP type and the presence of the > > paired substream for avoiding the crash. > > > > Reported-by: Jens Axboe <axboe(a)kernel.dk> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > Confirmed, it works. You can add my tested-by too. Thanks Takashi! Thanks for quick testing. Applied it now. Takashi

1 0

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 13:49:22 +0100, Jens Axboe wrote: > > On 2013-01-10 11:21, Takashi Iwai wrote: > > At Thu, 10 Jan 2013 09:23:48 +0100, > > Jens Axboe wrote: > >> > >> Hi, > >> > >> I have a USB DAC that I use for music. Just upgraded my workstation from > >> 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > >> an immediate oops. I have attached the picture. It's crashing here: > >> > >> (gdb) l *snd_usb_pcm_prepare+0x153 > >> 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > >> 465 snd_pcm_format_t pcm_format) > >> 466 { > >> 467 int i; > >> 468 int score = 0; > >> 469 > >> 470 if (fp->channels < 1) { > >> 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > >> 472 return 0; > >> 473 } > >> 474 > >> > >> 'fp' is clearly NULL. > > > > Hmm... it's a function called only from a single place in > > configure_sync_endpoint() and it's in list_for_each_entry(): > > > > list_for_each_entry(fp, &sync_subs->fmt_list, list) { > > int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, > > subs->cur_rate, subs->pcm_format); > > > > so it must be hardly NULL in normal situations. > > The only scenario I can imagine right now is that the whole structure > > is still uninitialized while it's called. If so, it's a race problem, > > and shouldn't be new to 3.8. A patch like below might paper over the > > problem although it's far from perfect. > > > > > >> Let me know if you want a bisect. > > > > Yeah, that'll be really helpful. > > Here it is, it's from the one introducing the audioformat lookup. > Confirmed that 3.8-rc3 with this backed out works fine, too. So should > be fairly confident in that result. Thanks! I'm slowly understanding what's happening there. Could you try the patch below? Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: usb-audio: Fix NULL dereference by access to non-existing substream The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for audioformat mismatch] introduced the correction of parameters to be set for sync EP. But since the new code assumes that the sync EP is always paired with the data EP of another direction, it triggers Oops when a device only with a single direction is used. This patch adds a proper check of sync EP type and the presence of the paired substream for avoiding the crash. Reported-by: Jens Axboe <axboe(a)kernel.dk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c659310..21c0001 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -511,6 +511,17 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) struct snd_usb_substream *sync_subs = &subs->stream->substream[subs->direction ^ 1]; + if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA || + !subs->stream) { + ret = snd_usb_endpoint_set_params(subs->sync_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + NULL); + } + /* Try to find the best matching audioformat. */ list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, -- 1.8.1

2 2

[alsa-devel] [PATCH] ASoC: dapm: Fix sense of regulator bypass mode
by Mark Brown 10 Jan '13

10 Jan '13

Enable bypass when the regulator is idle, not when it is in use. This is consistent with what the few existing users actually want. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc18d78..17dba18 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1023,7 +1023,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, if (SND_SOC_DAPM_EVENT_ON(event)) { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, true); + ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to bypass %s: %d\n", @@ -1033,7 +1033,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, return regulator_enable(w->regulator); } else { if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { - ret = regulator_allow_bypass(w->regulator, false); + ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, "ASoC: Failed to unbypass %s: %d\n", @@ -3039,6 +3039,14 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } + + if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { + ret = regulator_allow_bypass(w->regulator, true); + if (ret != 0) + dev_warn(w->dapm->dev, + "ASoC: Failed to unbypass %s: %d\n", + w->name, ret); + } break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP -- 1.7.10.4

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.8
by Mark Brown 10 Jan '13

10 Jan '13

The following changes since commit 9931faca02c604c22335f5a935a501bb2ace6e20: Linux 3.8-rc3 (2013-01-09 18:59:55 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-3.8-rc2 for you to fetch changes up to 49a170bcf28af5ef372cdbed7750ba8d4162bb0e: Merge remote-tracking branch 'asoc/fix/wm5100' into tmp (2013-01-10 12:22:30 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.8 Nothing terribly exciting here except for the DOUBLE_RANGE fix which just hadn't worked before, nobody noticed due to lack of use. ---------------------------------------------------------------- Axel Lin (5): ASoC: sta529: Fix update register bits in sta529_set_dai_fmt ASoC: wm2200: Fix setting dai format in wm2200_set_fmt ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt() ASoC: arizona: Do proper shift for setting AIF rate ASoC: cs42l52: Catch no-match case in cs42l52_get_clk Chuansheng Liu (2): ASoC: core: fix the memory leak in case of device_add() failure ASoC: core: fix the memory leak in case of remove_aux_dev() Daniel Mack (2): ASoC: cs4271: fix sparse warning ASoC: cs4271: fix property check Fabio Estevam (2): ASoC: soc-core: Remove unused 'ret' variable ASoC: sgtl5000: Fix maximum value for microphone gain MR.Swami.Reddy(a)ti.com (2): ASoC: lm49453: Fix adc, mic and sidetone volume ranges ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B Mark Brown (20): ASoC: core: Fix SOC_DOUBLE_RANGE() macros ASoC: arizona: Correct FLL source definitions ASoC: wm5102: Improve speaker enable performance ASoC: wm2200: Remove DSP B and left justified AIF modes ASoC: arizona: Remove DSP B and left justified AIF modes ASoC: wm5100: Remove DSP B and left justified formats ASoC: wm2000: Fix sense of speech clarity enable ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses Merge remote-tracking branch 'asoc/fix/arizona' into tmp Merge remote-tracking branch 'asoc/fix/core' into tmp Merge remote-tracking branch 'asoc/fix/cs4271' into tmp Merge remote-tracking branch 'asoc/fix/cs42l52' into tmp Merge remote-tracking branch 'asoc/fix/lm49453' into tmp Merge remote-tracking branch 'asoc/fix/pxa' into tmp Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmp Merge remote-tracking branch 'asoc/fix/sta529' into tmp Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmp Merge remote-tracking branch 'asoc/fix/wm2000' into tmp Merge remote-tracking branch 'asoc/fix/wm2200' into tmp Merge remote-tracking branch 'asoc/fix/wm5100' into tmp Mike Dunn (2): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset Patrick Lai (1): ASoC: pcm: allow backend hardware to be freed in pause state arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 + arch/arm/mach-pxa/pxa27x.c | 4 +- include/sound/cs4271.h | 2 +- include/sound/soc.h | 10 ++- sound/arm/pxa2xx-ac97-lib.c | 26 ++++++- sound/soc/codecs/arizona.c | 9 +-- sound/soc/codecs/arizona.h | 18 ++--- sound/soc/codecs/cs4271.c | 6 +- sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/lm49453.c | 106 ++++++++++----------------- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta529.c | 9 ++- sound/soc/codecs/wm2000.c | 4 +- sound/soc/codecs/wm2200.c | 8 +- sound/soc/codecs/wm5100.c | 6 -- sound/soc/codecs/wm5102.c | 48 +++++++++++- sound/soc/codecs/wm_adsp.c | 23 +++++- sound/soc/soc-core.c | 35 ++++++++- sound/soc/soc-pcm.c | 1 + 19 files changed, 201 insertions(+), 125 deletions(-)

2 1

[alsa-devel] [PATCH RFC 0/8] Use generic parser for HD-audio codec drivers
by Takashi Iwai 10 Jan '13

10 Jan '13

Hi, this is the next series of patches to convert various codec drivers to use the new generic parser code. Most of changes are obvious, but some places it gets tricky. As mentioned in another mail, the patches are found in sound-unstable git tree test/hda-migrate branch git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git Takashi

2 10

[alsa-devel] [PATCH] ASoC: ak4642: add Device Tree support
by Kuninori Morimoto 10 Jan '13

10 Jan '13

Support for loading the ak4642 codec module via devicetree. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/ak4642.txt | 17 ++++++++++ sound/soc/codecs/ak4642.c | 33 ++++++++++++++++++-- 2 files changed, 48 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt new file mode 100644 index 0000000..623d4e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.txt @@ -0,0 +1,17 @@ +AK4642 I2C transmitter + +This device supports I2C mode only. + +Required properties: + + - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" + - reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4648: ak4648@0x12 { + compatible = "asahi-kasei,ak4642"; + reg = <0x12>; + }; +}; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f0cdab..c78794d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -26,6 +26,7 @@ #include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> +#include <linux/of_device.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/initval.h> @@ -513,12 +514,31 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4648 = { }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct of_device_id ak4642_of_match[]; static int ak4642_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct device_node *np = i2c->dev.of_node; + const struct snd_soc_codec_driver *driver; + + driver = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4642_of_match, &i2c->dev); + if (of_id) + driver = of_id->data; + } else { + driver = (struct snd_soc_codec_driver *)id->driver_data; + } + + if (!driver) { + dev_err(&i2c->dev, "no driver\n"); + return -EINVAL; + } + return snd_soc_register_codec(&i2c->dev, - (struct snd_soc_codec_driver *)id->driver_data, - &ak4642_dai, 1); + driver, &ak4642_dai, 1); } static int ak4642_i2c_remove(struct i2c_client *client) @@ -527,6 +547,14 @@ static int ak4642_i2c_remove(struct i2c_client *client) return 0; } +static struct of_device_id ak4642_of_match[] __devinitconst = { + { .compatible = "asahi-kasei,ak4642", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4643", .data = &soc_codec_dev_ak4642}, + { .compatible = "asahi-kasei,ak4648", .data = &soc_codec_dev_ak4648}, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4642_of_match); + static const struct i2c_device_id ak4642_i2c_id[] = { { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 }, { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 }, @@ -539,6 +567,7 @@ static struct i2c_driver ak4642_i2c_driver = { .driver = { .name = "ak4642-codec", .owner = THIS_MODULE, + .of_match_table = ak4642_of_match, }, .probe = ak4642_i2c_probe, .remove = ak4642_i2c_remove, -- 1.7.9.5

2 1

Re: [alsa-devel] Oops in sound/usb/pcm.c:match_endpoint_audioformats() in current -git
by Takashi Iwai 10 Jan '13

10 Jan '13

At Thu, 10 Jan 2013 09:23:48 +0100, Jens Axboe wrote: > > Hi, > > I have a USB DAC that I use for music. Just upgraded my workstation from > 3.7-rc7 to 3.8-rc3 this morning, and when the DAC is switched on, I get > an immediate oops. I have attached the picture. It's crashing here: > > (gdb) l *snd_usb_pcm_prepare+0x153 > 0x161c is in snd_usb_pcm_prepare (sound/usb/pcm.c:470). > 465 snd_pcm_format_t pcm_format) > 466 { > 467 int i; > 468 int score = 0; > 469 > 470 if (fp->channels < 1) { > 471 snd_printdd("%s: (fmt @%p) no channels\n", __func__, fp); > 472 return 0; > 473 } > 474 > > 'fp' is clearly NULL. Hmm... it's a function called only from a single place in configure_sync_endpoint() and it's in list_for_each_entry(): list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, subs->cur_rate, subs->pcm_format); so it must be hardly NULL in normal situations. The only scenario I can imagine right now is that the whole structure is still uninitialized while it's called. If so, it's a race problem, and shouldn't be new to 3.8. A patch like below might paper over the problem although it's far from perfect. > Let me know if you want a bisect. Yeah, that'll be really helpful. Or, at least, could you try to copy sound/usb/* from 3.7 to 3.8 and see whether it's really a regression there? thanks, Takashi --- diff --git a/sound/usb/stream.c b/sound/usb/stream.c index ad181d5..3bfc564 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -81,8 +81,7 @@ static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) */ static void snd_usb_init_substream(struct snd_usb_stream *as, - int stream, - struct audioformat *fp) + int stream) { struct snd_usb_substream *subs = &as->substream[stream]; @@ -96,6 +95,13 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->speed = snd_usb_get_speed(subs->dev); snd_usb_set_pcm_ops(as->pcm, stream); +} + +static void snd_usb_add_substream_format(struct snd_usb_stream *as, + int stream, + struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; list_add_tail(&fp->list, &subs->fmt_list); subs->formats |= fp->formats; @@ -342,7 +348,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, err = snd_pcm_new_stream(as->pcm, stream, 1); if (err < 0) return err; - snd_usb_init_substream(as, stream, fp); + snd_usb_add_substream_format(as, stream, fp); return add_chmap(as->pcm, stream, subs); } @@ -370,7 +376,9 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, else strcpy(pcm->name, "USB Audio"); - snd_usb_init_substream(as, stream, fp); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_PLAYBACK); + snd_usb_init_substream(as, SNDRV_PCM_STREAM_CAPTURE); + snd_usb_add_substream_format(as, stream, fp); list_add(&as->list, &chip->pcm_list); chip->pcm_devs++;

1 0

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Make ebox44_table static
by Sachin Kamat 10 Jan '13

10 Jan '13

Fixes the following sparse warning: sound/usb/mixer_quirks.c:1209:23: warning: symbol 'ebox44_table' was not declared. Should it be static? Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/usb/mixer_quirks.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0422b13..15520de 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer) * are valid they presents mono controls as L and R channels of * stereo. So we provide a good mixer here. */ -struct std_mono_table ebox44_table[] = { +static struct std_mono_table ebox44_table[] = { { .unitid = 4, .control = 1, -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ALSA: hdspm: Fix wordclock status on AES32
by Andre Schramm 09 Jan '13

09 Jan '13

Use correct bitmask for AES32 cards to determine wordclock lock state, add missing bitmask for sync check and make output of the corresponding control and /proc coherent. Signed-off-by: Andre Schramm <andre.schramm(a)iosono-sound.com> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6e02e06..36e247f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); */ /* status */ #define HDSPM_AES32_wcLock 0x0200000 +#define HDSPM_AES32_wcSync 0x0100000 #define HDSPM_AES32_wcFreq_bit 22 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function HDSPM_bit2freq */ @@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) switch (hdspm->io_type) { case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_wcSync) - return 2; - else if (status & HDSPM_wcLock) - return 1; + if (status & HDSPM_AES32_wcLock) { + if (status & HDSPM_AES32_wcSync) + return 2; + else + return 1; + } return 0; break; @@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, unsigned int status; unsigned int status2; unsigned int timecode; + unsigned int wcLock, wcSync; int pref_syncref; char *autosync_ref; int x; @@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Status:\n"); + wcLock = status & HDSPM_AES32_wcLock; + wcSync = wcLock && (status & HDSPM_AES32_wcSync); + snd_iprintf(buffer, "Word: %s Frequency: %d\n", - (status & HDSPM_AES32_wcLock) ? "Sync " : "No Lock", + (wcLock) ? (wcSync ? "Sync " : "Lock") : "No Lock", HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF)); for (x = 0; x < 8; x++) { -- 1.7.1

3 4

[alsa-devel] [PATCH] Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"
by David Henningsson 09 Jan '13

09 Jan '13

This reverts commit 697c373e34613609cb5450f98b91fefb6e910588. The original patch was meant to remove clicking, but in fact caused even more clicking instead. Thanks to c4pp4 for doing most of the work with this bug. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- @c4pp4: Is this consistent with what you have done to reduce the clicking? Will you recommend upstream Linux to take the patch below? sound/pci/hda/patch_conexant.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 60890bf..dd798c3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM -static int conexant_suspend(struct hda_codec *codec) -{ - snd_hda_shutup_pins(codec); - return 0; -} -#endif - static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, .set_power_state = conexant_set_power, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = conexant_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; /* -- 1.7.9.5

2 7

Re: [alsa-devel] [3.6.9 -> 3.7.1 regression] sound: snd_hda_intel codec probing issue?
by Takashi Iwai 09 Jan '13

09 Jan '13

At Tue, 08 Jan 2013 21:25:21 +0100, Vincent Blut wrote: > > Le mardi 08 janvier 2013 à 13:56 +0100, Takashi Iwai a écrit : > > At Tue, 08 Jan 2013 13:28:55 +0100, > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Disable runtime D3 for Intel CPT & co > > > > We've got a few bug reports that the runtime D3 results in the dead > > HD-audio controller. It seems that the problem is in a deeper level > > than the sound driver itself, so as a temporal solution, disable the > > feature for these controllers again. > > > > Reported-by: Vincent Blut <vincent.debian(a)free.fr> > > Reported-by: Maurizio Avogadro <mavoga(a)gmail.com> > > Cc: <stable(a)vger.kernel.org> [v3.7] > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/hda_intel.c | 13 ++++++++----- > > 1 file changed, 8 insertions(+), 5 deletions(-) > > > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index cca8727..0b6aeba 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -573,9 +573,12 @@ enum { > > #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ > > > > /* quirks for Intel PCH */ > > -#define AZX_DCAPS_INTEL_PCH \ > > +#define AZX_DCAPS_INTEL_PCH_NOPM \ > > (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ > > - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME) > > + AZX_DCAPS_COUNT_LPIB_DELAY) > > + > > +#define AZX_DCAPS_INTEL_PCH \ > > + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) > > > > /* quirks for ATI SB / AMD Hudson */ > > #define AZX_DCAPS_PRESET_ATI_SB \ > > @@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci) > > static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > > /* CPT */ > > { PCI_DEVICE(0x8086, 0x1c20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* PBG */ > > { PCI_DEVICE(0x8086, 0x1d20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Panther Point */ > > { PCI_DEVICE(0x8086, 0x1e20), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, > > /* Lynx Point */ > > { PCI_DEVICE(0x8086, 0x8c20), > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > Well, my sound controller behaves nicely with the above quirk. > > Tested-by: Vincent Blut <vincent.debian(a)free.fr> OK, I queued the patch now. thanks, Takashi

1 0

Re: [alsa-devel] [PATCH v2 1/3] ASoC: tegra: add ac97 host driver
by Stephen Warren 08 Jan '13

08 Jan '13

On 01/08/2013 03:22 PM, Lucas Stach wrote: > Am Dienstag, den 08.01.2013, 15:10 -0700 schrieb Stephen Warren: >> On 01/04/2013 06:18 PM, Lucas Stach wrote: >>> This adds the driver for the Tegra 2x AC97 host controller. >>> diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c >>> +static inline void tegra20_ac97_start_playback(struct tegra20_ac97 *ac97) >>> +{ >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN); >>> + >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN | >>> + TEGRA20_AC97_CTRL_STM_EN, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN | >>> + TEGRA20_AC97_CTRL_STM_EN); >>> +} >> >> That sets both PCM_DAC_EN and STM_EN, but ... >> >>> +static inline void tegra20_ac97_stop_playback(struct tegra20_ac97 *ac97) >>> +{ >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_FIFO1_SCR, >>> + TEGRA20_AC97_FIFO_SCR_PB_QRT_MT_EN, 0); >>> + >>> + regmap_update_bits(ac97->regmap, TEGRA20_AC97_CTRL, >>> + TEGRA20_AC97_CTRL_PCM_DAC_EN, 0); >>> +} >> >> ... that only clears DAC_EN. Should it clear STM_EN too? >> > To be honest I have no idea what STM really is, seems to be some form of > packed format selection. If not set playback outputs only more or less > random noise. Only PCM_EN controls FIFO and DMA operations though, so > it's ok to just disable this to stop playback. I'd be tempted to just hard-code those bits on in probe() then, but it's not a big deal, so not worth spinning the patch for that. I guess I meant to say, Reviewed-by: Stephen Warren <swarren(a)nvidia.com>

1 0

[alsa-devel] [PATCH 0/3] ALSA: pxa27x: ac97 reset fixes
by Mike Dunn 08 Jan '13

08 Jan '13

AC97 reset is broken on the pxa27x. These patches fix it by addressing multiple problems: - cold reset fails because the GCR register is not set correctly - warm reset fails because some code used to work around a hardware bug is broken - the intent of the work-around code is obfuscated by unfortunate function naming This patch set replaces some earlier individual patches that had some problems. Thanks Robert and Igor. Changelog: v2: - don't specify the LPM flag DRIVE_HIGH in #define for the mfp config - patches 2 and 3 are combined into one patch - mfp level and direction are set only once, during driver initialization Mike Dunn (3): ALSA: pxa2xx: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset ALSA: pxa27x: rename pxa27x_assert_ac97reset() arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 ++ arch/arm/mach-pxa/pxa27x.c | 24 ++++++++++++------ sound/arm/pxa2xx-ac97-lib.c | 34 +++++++++++++++++++++----- 3 files changed, 46 insertions(+), 15 deletions(-) -- 1.7.8.6

4 9

[alsa-devel] [asoc:topic/adsp 4/4] sound/soc/codecs/wm_adsp.c:664:29: sparse: incorrect type in assignment (different modifiers)
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 25326bb263894a528288716901bd434372927844 [4/4] ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks sparse warnings: + sound/soc/codecs/wm_adsp.c:664:29: sparse: incorrect type in assignment (different modifiers) sound/soc/codecs/wm_adsp.c:664:29: expected struct wm_adsp_region *mem sound/soc/codecs/wm_adsp.c:664:29: got struct wm_adsp_region const * sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_load_coeff': sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type [enabled by default] vim +664 sound/soc/codecs/wm_adsp.c 2159ad93 Mark Brown 2012-10-11 648 case WMFW_NAME_TEXT: 2159ad93 Mark Brown 2012-10-11 649 case WMFW_INFO_TEXT: 2159ad93 Mark Brown 2012-10-11 650 break; 2159ad93 Mark Brown 2012-10-11 651 case WMFW_ABSOLUTE: 2159ad93 Mark Brown 2012-10-11 652 region_name = "register"; 2159ad93 Mark Brown 2012-10-11 653 reg = offset; 2159ad93 Mark Brown 2012-10-11 654 break; 25326bb2 Mark Brown 2013-01-08 655 25326bb2 Mark Brown 2013-01-08 656 case WMFW_ADSP1_DM: 25326bb2 Mark Brown 2013-01-08 657 case WMFW_ADSP1_ZM: 25326bb2 Mark Brown 2013-01-08 658 case WMFW_ADSP2_XM: 25326bb2 Mark Brown 2013-01-08 659 case WMFW_ADSP2_YM: 25326bb2 Mark Brown 2013-01-08 660 adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", 25326bb2 Mark Brown 2013-01-08 661 file, blocks, le32_to_cpu(blk->len), 25326bb2 Mark Brown 2013-01-08 662 type, le32_to_cpu(blk->id)); 25326bb2 Mark Brown 2013-01-08 663 25326bb2 Mark Brown 2013-01-08 @664 mem = wm_adsp_find_region(dsp, type); 25326bb2 Mark Brown 2013-01-08 665 if (!mem) { 25326bb2 Mark Brown 2013-01-08 666 adsp_err(dsp, "No base for region %x\n", type); 25326bb2 Mark Brown 2013-01-08 667 break; 25326bb2 Mark Brown 2013-01-08 668 } 25326bb2 Mark Brown 2013-01-08 669 25326bb2 Mark Brown 2013-01-08 670 reg = 0; 25326bb2 Mark Brown 2013-01-08 671 list_for_each_entry(alg_region, 25326bb2 Mark Brown 2013-01-08 672 &dsp->alg_regions, list) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [asoc:topic/adsp 4/4] sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 25326bb263894a528288716901bd434372927844 [4/4] ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:465:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm_adsp.c:480:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_load_coeff': sound/soc/codecs/wm_adsp.c:664:8: warning: assignment discards 'const' qualifier from pointer target type [enabled by default] vim +/const +664 sound/soc/codecs/wm_adsp.c 2159ad93 Mark Brown 2012-10-11 648 case WMFW_NAME_TEXT: 2159ad93 Mark Brown 2012-10-11 649 case WMFW_INFO_TEXT: 2159ad93 Mark Brown 2012-10-11 650 break; 2159ad93 Mark Brown 2012-10-11 651 case WMFW_ABSOLUTE: 2159ad93 Mark Brown 2012-10-11 652 region_name = "register"; 2159ad93 Mark Brown 2012-10-11 653 reg = offset; 2159ad93 Mark Brown 2012-10-11 654 break; 25326bb2 Mark Brown 2013-01-08 655 25326bb2 Mark Brown 2013-01-08 656 case WMFW_ADSP1_DM: 25326bb2 Mark Brown 2013-01-08 657 case WMFW_ADSP1_ZM: 25326bb2 Mark Brown 2013-01-08 658 case WMFW_ADSP2_XM: 25326bb2 Mark Brown 2013-01-08 659 case WMFW_ADSP2_YM: 25326bb2 Mark Brown 2013-01-08 660 adsp_dbg(dsp, "%s.%d: %d bytes in %x for %x\n", 25326bb2 Mark Brown 2013-01-08 661 file, blocks, le32_to_cpu(blk->len), 25326bb2 Mark Brown 2013-01-08 662 type, le32_to_cpu(blk->id)); 25326bb2 Mark Brown 2013-01-08 663 25326bb2 Mark Brown 2013-01-08 @664 mem = wm_adsp_find_region(dsp, type); 25326bb2 Mark Brown 2013-01-08 665 if (!mem) { 25326bb2 Mark Brown 2013-01-08 666 adsp_err(dsp, "No base for region %x\n", type); 25326bb2 Mark Brown 2013-01-08 667 break; 25326bb2 Mark Brown 2013-01-08 668 } 25326bb2 Mark Brown 2013-01-08 669 25326bb2 Mark Brown 2013-01-08 670 reg = 0; 25326bb2 Mark Brown 2013-01-08 671 list_for_each_entry(alg_region, 25326bb2 Mark Brown 2013-01-08 672 &dsp->alg_regions, list) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [asoc:topic/adsp 3/4] sound/soc/codecs/wm_adsp.c:464:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t'
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: 7265971f1bda27fdf97a19db3608ae15e2cdc7d2 [3/4] ASoC: wm_asdp: Validate sanity of algorithm count config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:464:3: warning: format '%d' expects argument of type 'int', but argument 4 has type 'size_t' [-Wformat] sound/soc/codecs/wm_adsp.c:479:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] vim +464 sound/soc/codecs/wm_adsp.c aab034fe Mark Brown 2012-10-26 448 aab034fe Mark Brown 2012-10-26 449 pos = sizeof(adsp2_id) / 2; aab034fe Mark Brown 2012-10-26 450 term = pos + ((sizeof(*adsp2_alg) * algs) / 2); aab034fe Mark Brown 2012-10-26 451 break; aab034fe Mark Brown 2012-10-26 452 aab034fe Mark Brown 2012-10-26 453 default: aab034fe Mark Brown 2012-10-26 454 BUG_ON(NULL == "Unknown DSP type"); aab034fe Mark Brown 2012-10-26 455 return -EINVAL; aab034fe Mark Brown 2012-10-26 456 } aab034fe Mark Brown 2012-10-26 457 aab034fe Mark Brown 2012-10-26 458 if (algs == 0) { aab034fe Mark Brown 2012-10-26 459 adsp_err(dsp, "No algorithms\n"); aab034fe Mark Brown 2012-10-26 460 return -EINVAL; aab034fe Mark Brown 2012-10-26 461 } aab034fe Mark Brown 2012-10-26 462 7265971f Mark Brown 2012-12-19 463 if (algs > 1024) { 7265971f Mark Brown 2012-12-19 @464 adsp_err(dsp, "Algorithm count %d excessive\n", algs); 7265971f Mark Brown 2012-12-19 465 print_hex_dump_bytes(dev_name(dsp->dev), DUMP_PREFIX_OFFSET, 7265971f Mark Brown 2012-12-19 466 buf, buf_size); 7265971f Mark Brown 2012-12-19 467 return -EINVAL; 7265971f Mark Brown 2012-12-19 468 } 7265971f Mark Brown 2012-12-19 469 aab034fe Mark Brown 2012-10-26 470 /* Read the terminator first to validate the length */ aab034fe Mark Brown 2012-10-26 471 ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); aab034fe Mark Brown 2012-10-26 472 if (ret != 0) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [asoc:topic/adsp 1/4] sound/soc/codecs/wm_adsp.c:446:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int'
by kbuild test robot 08 Jan '13

08 Jan '13

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/adsp head: 25326bb263894a528288716901bd434372927844 commit: aab034fe60460c0e352a64e94f2323edc12ff296 [1/4] ASoC: wm_adsp: Add support for parsing algorithms config: make ARCH=x86_64 allmodconfig All warnings: sound/soc/codecs/wm_adsp.c: In function 'wm_adsp_setup_algs': sound/soc/codecs/wm_adsp.c:446:3: warning: format '%zx' expects argument of type 'size_t', but argument 4 has type 'unsigned int' [-Wformat] vim +446 sound/soc/codecs/wm_adsp.c aab034fe Mark Brown 2012-10-26 430 } aab034fe Mark Brown 2012-10-26 431 aab034fe Mark Brown 2012-10-26 432 if (algs == 0) { aab034fe Mark Brown 2012-10-26 433 adsp_err(dsp, "No algorithms\n"); aab034fe Mark Brown 2012-10-26 434 return -EINVAL; aab034fe Mark Brown 2012-10-26 435 } aab034fe Mark Brown 2012-10-26 436 aab034fe Mark Brown 2012-10-26 437 /* Read the terminator first to validate the length */ aab034fe Mark Brown 2012-10-26 438 ret = regmap_raw_read(regmap, mem->base + term, &val, sizeof(val)); aab034fe Mark Brown 2012-10-26 439 if (ret != 0) { aab034fe Mark Brown 2012-10-26 440 adsp_err(dsp, "Failed to read algorithm list end: %d\n", aab034fe Mark Brown 2012-10-26 441 ret); aab034fe Mark Brown 2012-10-26 442 return ret; aab034fe Mark Brown 2012-10-26 443 } aab034fe Mark Brown 2012-10-26 444 aab034fe Mark Brown 2012-10-26 445 if (be32_to_cpu(val) != 0xbedead) aab034fe Mark Brown 2012-10-26 @446 adsp_warn(dsp, "Algorithm list end %zx 0x%x != 0xbeadead\n", aab034fe Mark Brown 2012-10-26 447 term, be32_to_cpu(val)); aab034fe Mark Brown 2012-10-26 448 aab034fe Mark Brown 2012-10-26 449 alg = kzalloc((term - pos) * 2, GFP_KERNEL); aab034fe Mark Brown 2012-10-26 450 if (!alg) aab034fe Mark Brown 2012-10-26 451 return -ENOMEM; aab034fe Mark Brown 2012-10-26 452 aab034fe Mark Brown 2012-10-26 453 ret = regmap_raw_read(regmap, mem->base + pos, alg, (term - pos) * 2); aab034fe Mark Brown 2012-10-26 454 if (ret != 0) { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

1 0

[alsa-devel] [PATCH v2] ASoC: mxs-saif: Use a signed integer for error value
by Fabio Estevam 08 Jan '13

08 Jan '13

saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Keep the attributions in the case of of_alias_get_id is not negative. sound/soc/mxs/mxs-saif.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675d..8f17094 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; + else + saif->id = ret; + /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +687,14 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; + else + saif->master_id = ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Factor out calculation of memory base addresses
by Mark Brown 08 Jan '13

08 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 +++++++++++++++++++++++++----- 1 file changed, 25 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6bff303..ed9a6c4 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -156,6 +156,26 @@ static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, return NULL; } +static unsigned int wm_adsp_region_to_reg(struct wm_adsp_region const *region, + unsigned int offset) +{ + switch (region->type) { + case WMFW_ADSP1_PM: + return region->base + (offset * 3); + case WMFW_ADSP1_DM: + return region->base + (offset * 2); + case WMFW_ADSP2_XM: + return region->base + (offset * 2); + case WMFW_ADSP2_YM: + return region->base + (offset * 2); + case WMFW_ADSP1_ZM: + return region->base + (offset * 2); + default: + WARN_ON(NULL != "Unknown memory region type"); + return offset; + } +} + static int wm_adsp_load(struct wm_adsp *dsp) { const struct firmware *firmware; @@ -283,27 +303,27 @@ static int wm_adsp_load(struct wm_adsp *dsp) case WMFW_ADSP1_PM: BUG_ON(!mem); region_name = "PM"; - reg = mem->base + (offset * 3); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_DM: BUG_ON(!mem); region_name = "DM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_XM: BUG_ON(!mem); region_name = "XM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP2_YM: BUG_ON(!mem); region_name = "YM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; case WMFW_ADSP1_ZM: BUG_ON(!mem); region_name = "ZM"; - reg = mem->base + (offset * 2); + reg = wm_adsp_region_to_reg(mem, offset); break; default: adsp_warn(dsp, -- 1.7.10.4

1 1

Re: [alsa-devel] [3.6.9 -> 3.7.1 regression] sound: snd_hda_intel codec probing issue?
by Takashi Iwai 08 Jan '13

08 Jan '13

At Fri, 28 Dec 2012 15:25:40 +0100, Vincent Blut wrote: > > Hi, > > Since I updated to Linux 3.7.1, listening to some audio/video bits > frequently cause the following: > > [ 7896.166946] hda-intel: azx_get_response timeout, switching to polling > mode: last cmd=0x020c0000 > [ 7897.173444] hda-intel: No response from codec, disabling MSI: last > cmd=0x020c0000 > [ 7898.179932] hda_intel: azx_get_response timeout, switching to > single_cmd mode: last cmd=0x020c0000 > [ 7898.179983] hda-codec: out of range cmd 0:0:20:400:fffff7ff > [ 9445.034371] plugin-containe[5873]: segfault at 7f44bb95e639 ip > 00007f44e454bca0 sp 00007f44c91165f8 error 4 in > libc-2.13.so[7f44e442c000+180000] > > It seems to be a codec probing failure (?). This is really fatal because > the sound become very choppy and can't recover until I reboot. > I'll try to play with 'probe_mask' kernel parameter to see if I can > narrow the correct codec slots! > > By the way I can't reproduce this on 3.6.9, so is there something that > changed in this area in 3.7.1? If it's new in 3.7, this could be a regression by runtime D3. Try to pass power_save_controller=0 option to snd-hda-intel module (or change it via sysfs dynamically). thanks, Takashi

4 10

[alsa-devel] [PATCH] ALSA: hda - print power state for AFG node in proc file
by David Henningsson 08 Jan '13

08 Jan '13

It seems useful, and power states are required for AFG nodes, so I see no reason not to print it. As a bonus, also print the AFG nid. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_proc.c | 2 ++ 1 file changed, 2 insertions(+) (Feel free to rephrase "State of AFG node 0x%02x:\n" if you like.) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 045e5d3..d6c8416 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -603,6 +603,8 @@ static void print_codec_info(struct snd_info_entry *entry, print_amp_caps(buffer, codec, codec->afg, HDA_INPUT); snd_iprintf(buffer, "Default Amp-Out caps: "); print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT); + snd_iprintf(buffer, "State of AFG node 0x%02x:\n", codec->afg); + print_power_state(buffer, codec, codec->afg); nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); if (! nid || nodes < 0) { -- 1.7.9.5

2 3

[alsa-devel] [PATCH] ASoC: mxs-saif: Use a signed integer for error value
by Fabio Estevam 08 Jan '13

08 Jan '13

saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675d..a920206 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,9 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +684,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH RESEND v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

2 3

[alsa-devel] [PATCH v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

2 5

Re: [alsa-devel] About Alsa arecord
by Raymond Yau 08 Jan '13

08 Jan '13

> Good morning! > Are you saying I should modify the source code of alsa-driver-1.0.25? >> >> > Hello! >> > There have 2 stereo ADCs support one stereo microphone and one legacy mixer recording simultaneously, said in the document of ALC888 Datasheet .I inputed 1KHz sine wave into every pins (Line1-R,Line1-L,MIC1-R,MIC1_L,MIC2-R,MIC2-L,LINE2-R,LINE2-L) of ALC888 by signal genrator,and used command 'arecord -c 4 -f S16_LE -r 44100 >test.wav' recording in fedora14.but ALL audio data are recorded in channel 1,2.No any data in channel 3,4.Why?How can I record every pins audio data in each channel? >> >> In alc_auto_create_input_ctls , add the connection from node 0x0b of mixer ADC as input source similar to those pin complex >> >> Those Mic/line/CD playback volume and switches of node 0x0b affect the signal of the new input source ”mixer” > > Yes, also need a switch which mute the input from 0x0b for all output jacks to prevent feedback if using mic the current hda driver only support stereo capture and you have to implement multi channel capturing similar to multi channel playback in hda_codec.c snd_hda_multi_in_analog_prepare for (i = 1; i < num_adcs; i++) { snd_hda_codec_setup_stream(codec, adc_nids[i], stream_tag, i * 2, format); or try multi plugin of two stereo capture devices

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses
by Mark Brown 07 Jan '13

07 Jan '13

Otherwise we won't run correctly on systems that require this for larger data transfers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5d8bb1d..a765458 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; + void *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - ret = regmap_raw_write(regmap, reg, region->data, + buf = kmemdup(region->data, le32_to_cpu(region->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + + ret = regmap_raw_write(regmap, reg, buf, le32_to_cpu(region->len)); + + kfree(buf); + if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", -- 1.7.10.4

1 0

[alsa-devel] [PATCH 0/4] ALSA: pxa27x: ac97 reset fixes
by Mike Dunn 07 Jan '13

07 Jan '13

AC97 reset is broken on the pxa27x. These patches fix it by addressing multiple problems: - cold reset fails because the GCR register is not set correctly - warm reset fails because some code used to work around a hardware bug is broken - the intent of the work-around code is obfuscated by unfortunate function naming This patch set replaces some earlier individual patches that had some problems. Thanks Robert and Igor. Mike Dunn (4): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset bug work-around code ALSA: pxa27x: ac97 controller driver requests gpio ALSA: pxa27x: rename pxa27x_assert_ac97() arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 ++ arch/arm/mach-pxa/pxa27x.c | 38 +++++++++++++++++++------- sound/arm/pxa2xx-ac97-lib.c | 26 +++++++++++++----- 3 files changed, 50 insertions(+), 17 deletions(-) -- 1.7.8.6

4 21

[alsa-devel] [PATCH v2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..71ae23d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,6 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } +/* update mute-LED according to the speaker mute state via mic1 VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + /* update mute-LED according to the speaker mute state via mic2 VREF pin */ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { @@ -6043,6 +6067,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6196,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6244,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 1

[alsa-devel] Alsa snapshot directory layout changed
by David Henningsson 07 Jan '13

07 Jan '13

I have a script that picks daily snapshots from ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-snapshot.tar.gz ...and I just noticed that the directory layout has changed, so what was "alsa-driver" is now "alsa-driver/alsa". Is this expected, i e, should I update my script to follow this new layout, or was it a mistake? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 3

[alsa-devel] [PATCH] ALSA: pxa2xx: fix ac97 cold reset for pxa27x
by Mike Dunn 07 Jan '13

07 Jan '13

Currently, ac97 reset is broken on PXA27x. Through trial-and-error (the pxa270 developer's manual is mostly incoherent on the topic of ac97 reset), I fixed it by setting the WARM_RST bit in the GCR register at the end of the cold reset, and then skipping the warm reset if the link is already up and running. It appears that setting the WARM_RST bit is a necessary final step during cold reset. I think that the PXA25x and PXA3xx may currently be working correctly because WARM_RST is set within the warm reset routine, and the codec drivers always follow a cold reset with a warm reset during their initialization. so this combination effectively completes a cold reset. This doesn't work on the PXA27x because its warm reset routine contains additional code for working around a hardware bug in the warm reset sequence, which causes the reset sequence to fail. I only have a PXA27x platform for testing, so this patch only affects the PXA27x. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- As an aside... I don't understand how always following a cold reset with a warm reset (as the codec drivers do) makes sense. Unless I'm mistaken, a warm reset can only occur on a link that's currently down. All the timing diagrams I've seen for reset (warm and cold) show the link down (BITCLK stopped) at the start of the reset sequence. The SYNC line is used to effect the warm reset, and SYNC is already pulsing every frame on a link that's up and running, so it seems physically impossible to do a warm reset on a link that's currently up. But I'm no ac97 expert, so I may just be embarassing myself. Any insight gratefully received. Thanks! sound/arm/pxa2xx-ac97-lib.c | 15 +++++++++++++++ 1 files changed, 15 insertions(+), 0 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fc0ae9..aa242ff 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -138,6 +138,10 @@ static inline void pxa_ac97_warm_pxa27x(void) { gsr_bits = 0; + /* don't attempt warm reset if link is up */ + if (GSR & (GSR_SCR | GSR_PCR)) + return; + /* warm reset broken on Bulverde, so manually keep AC97 reset high */ pxa27x_assert_ac97reset(reset_gpio, 1); udelay(10); @@ -148,6 +152,8 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { + unsigned int timeout; + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -159,6 +165,15 @@ static inline void pxa_ac97_cold_pxa27x(void) clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); + + /* + * Setting the WARM_RST bit is a necessary part of a cold reset, at + * least on the buggy and poorly documented PXA27x ac97 controller. + */ + GCR |= GCR_WARM_RST; + timeout = 100; /* wait for the codec-ready bit to be set */ + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); } #endif -- 1.7.8.6

3 6

[alsa-devel] [PATCH 1/2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) I know this is a copy-paste, but I didn't really find a better and more condensed way to do it. Perhaps if we get a third variation it would make sense to write a more generic version. Also, Alsa-info is available at https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1096789/+attachment/34… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..2e6803e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,7 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } -/* update mute-LED according to the speaker mute state via mic2 VREF pin */ +/* update mute-LED according to the speaker mute state via mic VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -6043,6 +6066,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6195,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

2 4

[alsa-devel] About Alsa arecord
by FF 07 Jan '13

07 Jan '13

Hello! There have 2 stereo ADCs support one stereo microphone and one legacy mixer recording simultaneously, said in the document of ALC888 Datasheet .I inputed 1KHz sine wave into every pins (Line1-R,Line1-L,MIC1-R,MIC1_L,MIC2-R,MIC2-L,LINE2-R,LINE2-L) of ALC888 by signal genrator,and used command 'arecord -c 4 -f S16_LE -r 44100 >test.wav' recording in fedora14.but ALL audio data are recorded in channel 1,2.No any data in channel 3,4.Why?How can I record every pins audio data in each channel? Thank you!Best regards.

2 1

[alsa-devel] Roland-Boss RC-300 support
by Jean-Samuel Reynaud 06 Jan '13

06 Jan '13

Hi All, I have a Boss RC-300 loop station with an USB connector. The windows driver add an audio device but it was not working under linux. So please find attached, the patch for supporting audio in this device. Note that midi is not working yet but I'm working on it... Regards,

2 1

Re: [alsa-devel] Missing exactly 3 of 8 audio packets?
by Daniel Griscom 06 Jan '13

06 Jan '13

Dear Daniel (M) and Clemens, I'd like to conclude this thread if I could. It seems clear that configuring our USB audio card's descriptor with a bInterval value of 1 (a packet every microframe) was too aggressive, and was part of what caused the packet stuttering issue. (Right?) We almost certainly should increase bInterval; I'll try 2, and if that seems stable will probably go to 3 (4 microframes) for some margin. However, I'm still concerned that the symptoms don't sound like a simple starvation issue, where I'd expect to have a missing frame every once in a while. Instead, we were seeing reliable operation interspersed with periods where lots of packets were being consistently omitted. To me, this suggests that there's some other issue going on, either in the USB stack or in the USB hardware itself. Given that, I'm worried that increasing bInterval will just move the symptoms around, making it harder to duplicate but not truly becoming robust. Any thoughts on this? Should I try chasing the issue into the Linux USB development mailing list, or will they just say "set bInterval to a larger value!"? Thanks, Dan -- Daniel T. Griscom griscom(a)suitable.com Suitable Systems http://www.suitable.com/ 1 Centre Street, Suite 204 (781) 665-0053 Wakefield, MA 01880-2400

2 1

Re: [alsa-devel] [PATCH v3 0/5] ARM: OMAP4+: HDMI: Update platform devices for audio
by Ricardo Neri 05 Jan '13

05 Jan '13

Hi Mark, Tomi, Just wondering if you had a chance to take a look at these patches. Thanks! Ricardo On 12/17/2012 07:30 PM, Ricardo Neri wrote: > Hi Mark, Tomi, Liam, Tony, > > This set aims to be the version 3 of my previous submission[1] and aims to > address the comments that Mark and Tomi kindly provided on such submission. > > The creation of the platform device for the HDMI audio interface from within the > OMAPDSS HDMI driver that was previously submitted[2] is resubmitted to be > complemented with code to relocate to arch/arm/mach-omap2/display.c the > creation of the platform devices for the HDMI ASoC codec and card drivers. This > series does not break the HDMI audio functionality in any patch. > > Also, the names of the platform devices are changed to give them more logical > and more descriptive names. > > As the commit > > commit 14840b9a83c6a56629db2ba0ec247503e975f143 > Author: Ricardo Neri <ricardo.n...(a)ti.com> > Date: Tue Nov 6 00:19:17 2012 -0600 > > OMAPDSS: HDMI: Create platform device for audio support > > is reverted in Tomi's git://gitorious.org/linux-omap-dss2/linux.git master > branch, this series applies cleanly. > > Changes from v1: > *Put in a single series all the patches related to platform device updates. > *Now HDMI audio works correctly in every patch. > *Remove reference to the TPD12S015 HDMI companion chip as the ASoC drivers > are not aware of this and other chips could be used in the future. > > Changes from v2: > *Split in two patches the renaming and the relocation of the platform devices. > *Create a separate patch to pass only the address offset of the DMA data port > to the audio interface platform device. > *Keep the name hdmi-audio-codec to not refer to explicitly to OMAP. The codec > can be made generic in a different patch series submitted to alsa-devel. > > BR, > > Ricardo > > > [1]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg80785.html > [2]. http://www.mail-archive.com/linux-omap@vger.kernel.org/msg79795.html > > Ricardo Neri (5): > ARM: OMAP4+: HDMI: Rename platform devices for ASoC drivers > ARM: OMAP4: Assign IDs to DSS HDMI devices > ARM4: OMAP4+: HDMI: Relocate devices for audio codec and card > ARM: OMAP4+: HDMI: Relocate the device for audio interface > ARM: OMAP4+: HDMI: Refine the DMA port resource for audio > > arch/arm/mach-omap2/board-4430sdp.c | 9 ++--- > arch/arm/mach-omap2/board-omap4panda.c | 9 ++--- > arch/arm/mach-omap2/devices.c | 31 ---------------- > arch/arm/mach-omap2/display.c | 31 ++++++++++++++++ > drivers/video/omap2/dss/hdmi.c | 62 ++++++++++++++++++++++++++++++++ > sound/soc/omap/omap-hdmi-card.c | 4 +-- > sound/soc/omap/omap-hdmi.c | 5 ++- > sound/soc/omap/omap-hdmi.h | 2 -- > 8 files changed, 103 insertions(+), 50 deletions(-) > > -- > 1.7.10.4 >

2 2

[alsa-devel] [PATCH] ASoC: wm2000: Fix sense of speech clarity enable
by Mark Brown 04 Jan '13

04 Jan '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1cbe88f..12bcae6 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); if (wm2000->speech_clarity) - ret &= ~WM2000_SPEECH_CLARITY; - else ret |= WM2000_SPEECH_CLARITY; + else + ret &= ~WM2000_SPEECH_CLARITY; wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: arizona: Allow runtime reconfiguration of the output mode
by Mark Brown 04 Jan '13

04 Jan '13

Some systems use external analogue switches to connect more analogue devices to the CODEC than are supported by the device. In some systems this requires changing the switched output from single ended to differential mode dynamically at runtime. Add a new function arizona_set_output_mode() to support this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 34 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 3 +++ 2 files changed, 37 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7724b4b..6d98cf4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1176,6 +1176,40 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, } EXPORT_SYMBOL_GPL(arizona_init_fll); +/** + * arizona_set_output_mode - Set the mode of the specified output + * + * @codec: Device to configure + * @output: Output number + * @diff: True to set the output to differential mode + * + * Some systems use external analogue switches to connect more + * analogue devices to the CODEC than are supported by the device. In + * some systems this requires changing the switched output from single + * ended to differential mode dynamically at runtime, an operation + * supported using this function. + * + * Most systems have a single static configuration and should use + * platform data instead. + */ +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff) +{ + unsigned int reg, val; + + if (output < 1 || output > 6) + return -EINVAL; + + reg = ARIZONA_OUTPUT_PATH_CONFIG_1L + (output - 1) * 8; + + if (diff) + val = ARIZONA_OUT1_MONO; + else + val = 0; + + return snd_soc_update_bits(codec, reg, ARIZONA_OUT1_MONO, val); +} +EXPORT_SYMBOL_GPL(arizona_set_output_mode); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 68dfae5..697ff6f 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -213,4 +213,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, extern int arizona_init_dai(struct arizona_priv *priv, int dai); +int arizona_set_output_mode(struct snd_soc_codec *codec, int output, + bool diff); + #endif -- 1.7.10.4

1 0

Re: [alsa-devel] draft -- ALSA firewire + digi003
by Robin Gareus 04 Jan '13

04 Jan '13

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Clemens, Hi ALSA-devs, I'm working with Damien Zammit on the Digidesign-003 driver which is based on your alsa-kprivate dice code. We were able to work out most of the issues so far but are now stuck with timing problems. More specifically: the jitter of async message timing is quite large and the AMDTP ISO stream has regular drop-outs. Comparing 1394 data dumps, OSX sends async messages at regular 125 (+-5us) intervals. On GNU/Linux we've seen a delta -40us, + 120us. For async messages this is not a big deal. But.. Just playing audio via aplay: there are short dropouts every couple of seconds but otherwise the devices plays fine indefinitely. Interestingly, when running cyclictest while playing sound: there are no dropouts but the stream drops after ~10-15 seconds. Any hints why that may be or how to debug the issue further? In order to rule out kernel related issues (3.0.0 is quite old), we've rebased the work (and alsa-kprivate's firewire-kernel-streaming branch) onto alsa-driver.git. The issue remains the same with 3.2.0-4-rt Alas, neither of us has detailed knowledge nor specs for AMDTP, not to mention in-depth knowledge about Linux' firewire stack timings. Any help or leads are appreciated. As we believe we are close to a fully functional driver for the 003, we are very keen to resolve the following two separate issues: 1) The timing problem we have observed with AMDTP over firewire. 2) A framework for the capture stream in ALSA firewire. The rebased driver code is at git://gareus.org/alsa-driver.git http://gareus.org/gitweb/?p=alsa-driver.git The broken-out digi003 driver is available from: https://github.com/zamaudio/snd-firewire-003 https://github.com/x42/003amdtp 1394 bus data dumps of both OSX as well as Linux can be found at http://www.zamaudio.com/?p=715 Cheers! robin -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJQ5t9CAAoJEKCQvOAs9X8ELbsP/Re0wxZKU6v2rbBVhV/FC6zr JYmx/jAsmoC7vtVZMKE4gGcCreszX58oU7RG323qR9iKSEgXeqtuThZSTT6l3N39 KpHW4T094FyQxmvxRYrZnRlHgWexvLfJT/YiuTUX50L/CKgS75ajRfLqADZaHSy5 Oc+CEy50Jmoi/9/dTuM0pVPVvt5APFqLyAisw1/5wzhYMTHF0mBByn5AHXBp7bIl SVg7IoJuTNd4dYgo3PtKw9OqhjMGZ1g5DNHA9OaE7qY12r3TMMKm7Sq2iWS4HQoq RMQLOVtyMOVUKyddK8ubWi0Cm2lYjHUXUNSLyZLcdWYTWbk5eWo6Ze83Tbl33h4T ngiWt+DWpqKpm+U0UQDOU6p+DSAosKHsVcPPPkYVkwXKAkHctXKlPUst5NHEhhK3 5mbVQECHE9+BBy9qY2eimDZ/p3+5C2kIMofDga6jiuFWfHMqKp30uacUHwP4xY38 jHLaTmpX7VNwNYGnFLvGve709ejNjL8ByFKrUKpV1nAMTO4QP4SipHuApBF3BWS7 hAGBD6mKQAPl6AhvuJaxtCdTQc/tkAF0Ep1LClthsHnyhqPVCX+L5sb/z7vVmCkp cFxGXJq6TujcW4kkKPfZCNsnHqea8lgY6tEonc8aRmcIZvF+L4Yp+p4/r9OxL6ma xIIgIVDp1X9HxVayBDWL =lVsY -----END PGP SIGNATURE-----

1 0

[alsa-devel] [PATCH 1/2] ASoC: arizona: Remove DSP B and left justified AIF modes
by Mark Brown 04 Jan '13

04 Jan '13

These are not supported. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a4d7fb9..7724b4b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -503,15 +503,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mode = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mode = 1; - break; case SND_SOC_DAIFMT_I2S: mode = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mode = 3; - break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); -- 1.7.10.4

1 1

Re: [alsa-devel] sound: usb-audio: [PATCH]x2 full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 04 Jan '13

04 Jan '13

On 23 December 2012 14:36, Damien Zammit <damien.zammit(a)gmail.com> wrote: > I have fixed both the kernel panic and the broken pipe issue with midi. > I believe this completes the patch, I just don't know how to submit it to you. Please find attached the patch you accepted previously: (Version 5) Plus an incremental patch (Version 5-2) ***Please ignore patch (Version 6)*** I hope this is the easiest way for you to review it. It compiles, runs, and works on the hardware now without errors. Thanks, Damien

2 6

Re: [alsa-devel] [PATCH] sound: Add support for Creative BT-D1 via usb sound quirks
by Takashi Iwai 04 Jan '13

04 Jan '13

At Thu, 03 Jan 2013 12:59:07 +0100, Alexander Schremmer wrote: > > On 03.01.2013 10:58, Takashi Iwai wrote: > > > Thanks for the patch. The changes look OK, but could you fix trivial > > coding issues reported by scripts/checkpatch.pl and resend? > > Oops, my sending workflow ate the tabs, here it is again (hopefully now > working correctly): > > > From: Alexander Schremmer <alex(a)alexanderweb.de> > > Support the Creative BT-D1 Bluetooth USB audio device. Before this > patch, Linux had trouble finding the correct USB descriptors and bailed > out with these messages: > > no or invalid class specific endpoint descriptor > > Now it still prints these messages on hotplug: > > snd-usb-audio: probe of ...:1.0 failed with error -5 > snd-usb-audio: probe of ...:1.2 failed with error -5 > snd-usb-audio: probe of ...:1.3 failed with error -5 > > But the device works correctly, including the HID support. > > The patch is diff'ed against 3.8-rc1 but should apply to older kernels > as well. > > Signed-off-by: Alexander Schremmer <alex(a)alexanderweb.de> Thanks, applied. Takashi > > --- > diff -uprN linux-3.8-rc1/sound/usb/quirks-table.h > linux-3.8-rc1-mod/sound/usb/quirks-table.h > --- linux-3.8-rc1/sound/usb/quirks-table.h 2012-12-22 02:19:00.000000000 > +0100 > +++ linux-3.8-rc1-mod/sound/usb/quirks-table.h 2012-12-26 > 11:41:28.000000000 +0100 > @@ -50,6 +50,28 @@ > } > }, > > +{ > + /* Creative BT-D1 */ > + USB_DEVICE(0x041e, 0x0005), > + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { > + .ifnum = 1, > + .type = QUIRK_AUDIO_FIXED_ENDPOINT, > + .data = &(const struct audioformat) { > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels = 2, > + .iface = 1, > + .altsetting = 1, > + .altset_idx = 1, > + .endpoint = 0x03, > + .ep_attr = USB_ENDPOINT_XFER_ISOC, > + .attributes = 0, > + .rates = SNDRV_PCM_RATE_CONTINUOUS, > + .rate_min = 48000, > + .rate_max = 48000, > + } > + } > +}, > + > /* Creative/Toshiba Multimedia Center SB-0500 */ > { > USB_DEVICE(0x041e, 0x3048), >

1 0