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

Re: [alsa-devel] ASoC: rsnd: care DMA slave channel name for DT
by Dan Carpenter 11 Jun '14

11 Jun '14

Hello Kuninori Morimoto, The patch 199e7688bdf7: "ASoC: rsnd: care DMA slave channel name for DT" from May 22, 2014, leads to the following static checker warning: sound/soc/sh/rcar/core.c:318 rsnd_dma_of_name() error: buffer overflow 'mod' 4 <= 4 sound/soc/sh/rcar/core.c 278 struct rsnd_mod *mod[MOD_MAX]; 279 struct rsnd_mod *src_mod, *dst_mod; 280 int i, index; 281 282 283 for (i = 0; i < MOD_MAX; i++) 284 mod[i] = NULL; 285 286 /* 287 * in play case... 288 * 289 * src -> dst 290 * 291 * mem -> SSI 292 * mem -> SRC -> SSI 293 * mem -> SRC -> DVC -> SSI I don't understand what this comment means. 294 */ 295 mod[0] = NULL; /* for "mem" */ We memset everything to NULL in the loop above so this isn't needed. 296 index = 1; 297 for (i = 1; i < MOD_MAX; i++) { 298 if (!src) { 299 mod[i] = ssi; 300 break; 301 } else if (!dvc) { 302 mod[i] = src; 303 src = NULL; 304 } else { 305 mod[i] = dvc; 306 dvc = NULL; 307 } 308 309 if (mod[i] == this) 310 index = i; In theory, index can be MOD_MAX - 1. 311 } 312 313 if (is_play) { 314 src_mod = mod[index - 1]; 315 dst_mod = mod[index]; 316 } else { 317 src_mod = mod[index]; 318 dst_mod = mod[index + 1]; So then it complains that mod[] has only MOD_MAX elements so we're one space past the end of the array. Probably the way this is called, there is something to prevent it? 319 } regards, dan carpenter

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[alsa-devel] [PATCH 1/1] SND/PCI/HDA: add __init to snd_hda_add_codec_preset
by Fabian Frederick 10 Jun '14

10 Jun '14

snd_hda_add_codec_preset is only called by __init functions: __init patch_via_init __init patch_sigmatel_init __init patch_si3054_init __init patch_realtek_init __init patch_hdmi_init __init patch_conexant_init __init patch_cmedia_init __init patch_cirrus_init __init patch_ca0132_init __init patch_ca0110_init __init patch_analog_init Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_codec.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4c20277..5770c5c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -77,7 +77,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { static DEFINE_MUTEX(preset_mutex); static LIST_HEAD(hda_preset_tables); -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) +int __init snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) { mutex_lock(&preset_mutex); list_add_tail(&preset->list, &hda_preset_tables); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5825aa1..02a9123 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -182,7 +182,7 @@ struct hda_codec_preset_list { }; /* initial hook */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); +int __init snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); /* ops set by the preset patch */ -- 1.8.4.5

2 2

[alsa-devel] [PATCH 1/2] misc: atmel-ssc: support to read ssc version
by Bo Shen 10 Jun '14

10 Jun '14

Read SSC IP version to check whether it supports frame sync length extension feature. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- drivers/misc/atmel-ssc.c | 7 +++++++ include/linux/atmel-ssc.h | 3 +++ 2 files changed, 10 insertions(+) diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index 22de137..9f49bcb 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -136,6 +136,7 @@ static int ssc_probe(struct platform_device *pdev) struct resource *regs; struct ssc_device *ssc; const struct atmel_ssc_platform_data *plat_dat; + u32 version; ssc = devm_kzalloc(&pdev->dev, sizeof(struct ssc_device), GFP_KERNEL); if (!ssc) { @@ -173,8 +174,14 @@ static int ssc_probe(struct platform_device *pdev) clk_prepare_enable(ssc->clk); ssc_writel(ssc->regs, IDR, -1); ssc_readl(ssc->regs, SR); + version = ssc_readl(ssc->regs, VERSION); clk_disable_unprepare(ssc->clk); + if (version >= 0x300) + ssc->has_fslen_ext = 1; + else + ssc->has_fslen_ext = 0; + ssc->irq = platform_get_irq(pdev, 0); if (!ssc->irq) { dev_dbg(&pdev->dev, "could not get irq\n"); diff --git a/include/linux/atmel-ssc.h b/include/linux/atmel-ssc.h index 571a12e..9ee1b68 100644 --- a/include/linux/atmel-ssc.h +++ b/include/linux/atmel-ssc.h @@ -19,6 +19,7 @@ struct ssc_device { int user; int irq; bool clk_from_rk_pin; + int has_fslen_ext; }; struct ssc_device * __must_check ssc_request(unsigned int ssc_num); @@ -258,6 +259,8 @@ void ssc_free(struct ssc_device *ssc); #define SSC_IMR_TXSYN_SIZE 1 #define SSC_IMR_TXSYN_OFFSET 10 +#define SSC_VERSION 0x000000fc + /* SSC PDC Receive Pointer Register */ #define SSC_PDC_RPR 0x00000100 -- 1.8.5.2

2 3

[alsa-devel] [PATCH] ALSA: hda - Add quirk for ABit AA8XE
by David Henningsson 10 Jun '14

10 Jun '14

Bios does not set up the pin config default correctly (everything is set to zero). Reporter claims that 6stack-dig and 6stack-automute solve the problem. Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05 BugLink: https://bugs.launchpad.net/bugs/1319291 Reported-by: Stefano Statuti <stefano.statuti(a)hotmail.it> 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 d77745a..9c095fa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1427,6 +1427,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS W5A", ALC880_FIXUP_ASUS_W5A), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), + SND_PCI_QUIRK(0x147b, 0x1045, "ABit AA8XE", ALC880_FIXUP_6ST_AUTOMUTE), SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), -- 1.9.1

2 1

[alsa-devel] Fwd: Re: Patch for power on headphone amp on Realtek 883
by asd 10 Jun '14

10 Jun '14

Hi Here is content; #uname -a Linux marian1000.go.ro 3.14-1-amd64 #1 SMP Debian 3.14.4-1 (2014-05-13) x86_64 GNU/Linux #cat /proc/asound/Intel/codec#0 Codec: Realtek ALC883 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10ec0883 Subsystem Id: 0x1043829f Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC883 Analog", type="Audio", device=0 Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=2 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=4 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=8, channel=6 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC883 Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC883 Analog", type="Audio", device=0 Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x10 0x10] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC883 Alt Analog", type="Audio", device=2 Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=3, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Front Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Front Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Rear Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Side Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [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="Line Out Front 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 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=05, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [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="Line Out 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 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=06, enabled=1 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Center Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Switch", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Control: name="Line Out CLFE 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 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=07, enabled=1 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Side Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Line Out Side 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 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=08, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Rear Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Rear Mic 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: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19840: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=03, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Front Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Front Mic 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: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x5, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Line Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Line 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: [0x80 0x80] Pincap 0x0000173e: IN OUT HP Detect Trigger Vref caps: HIZ 50 GRD 80 Pin Default 0x0181304f: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 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="Front 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 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x4015e601: [N/A] Speaker at Ext N/A Conn = Optical, Color = White DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x01441130: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1f 0x1f] Connection: 2 0x25 0x0b

2 1

[alsa-devel] [PATCH v2 0/5] As the sama5d3xek and at91sam9n12ek have switch to CCF,
by Bo Shen 10 Jun '14

10 Jun '14

in order to support audio, also switch it to CCF. This patch series based on next-2014-0604 tag. Changes in v2: - wm8904: Remove the mclk checking when try to enable or disable it, as it is checked when required. - dt: List the clock-names, which need by driver. Bo Shen (5): ASoC: atmel_wm8904: switch to CCF ASoC: wm8904: switch to CCF dt: wm8904: add device tree binding document ARM: atmel: sama5d3xek: switch sound to CCF ARM: atmel: at91sam9n12ek: switch sound to CCF Documentation/devicetree/bindings/sound/wm8904.txt | 33 ++++++++++++++ arch/arm/boot/dts/at91sam9n12ek.dts | 2 + arch/arm/boot/dts/sama5d3xmb.dtsi | 2 + sound/soc/atmel/atmel_wm8904.c | 50 ---------------------- sound/soc/codecs/wm8904.c | 11 +++++ 5 files changed, 48 insertions(+), 50 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt -- 1.8.5.2

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[alsa-devel] [PATCH] sound: soc: pxa2xx-ac97: prepare and unprepare the clocks
by Robert Jarzmik 09 Jun '14

09 Jun '14

Add the clock prepare and unprepare call to the driver initialization phase. This will remove a warning once the PXA architecture is migrated to the clock infrastructure. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/arm/pxa2xx-ac97-lib.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 66de90e..39c3969 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -152,9 +152,9 @@ static inline void pxa_ac97_cold_pxa27x(void) gsr_bits = 0; /* PXA27x Developers Manual section 13.5.2.2.1 */ - clk_enable(ac97conf_clk); + clk_prepare_enable(ac97conf_clk); udelay(5); - clk_disable(ac97conf_clk); + clk_disable_unprepare(ac97conf_clk); GCR = GCR_COLD_RST | GCR_WARM_RST; } #endif @@ -299,14 +299,14 @@ static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) int pxa2xx_ac97_hw_suspend(void) { GCR |= GCR_ACLINK_OFF; - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - clk_enable(ac97_clk); + clk_prepare_enable(ac97_clk); return 0; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); @@ -368,7 +368,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk; } - ret = clk_enable(ac97_clk); + ret = clk_prepare_enable(ac97_clk); if (ret) goto err_clk2; @@ -403,7 +403,7 @@ void pxa2xx_ac97_hw_remove(struct platform_device *dev) clk_put(ac97conf_clk); ac97conf_clk = NULL; } - clk_disable(ac97_clk); + clk_disable_unprepare(ac97_clk); clk_put(ac97_clk); ac97_clk = NULL; } -- 2.0.0.rc2

2 3

[alsa-devel] [PATCH] ASoC: sigmadsp: Split regmap and I2C support into separate modules
by Lars-Peter Clausen 09 Jun '14

09 Jun '14

When the SigmaDSP module is built-in, but the I2C core is build as a module we'll get a undefined reference: sound/built-in.o: In function `sigma_action_write_i2c': :(.text+0x5d8d4): undefined reference to `i2c_master_send' This can happen if a audio driver that is using the regmap SigmaDSP interface is built into the kernel, but core I2C support is build as a module. To fix this split the SigmaDSP module into three modules, one module providing the core infrastructure and two small modules implementing the regmap and I2C interfaces. This allows e.g. the core infrastructure and regmap support to be built into the kernel while I2C support can still be build as a module. Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support") Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 12 +++++-- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/sigmadsp-i2c.c | 35 ++++++++++++++++++++ sound/soc/codecs/sigmadsp-regmap.c | 36 +++++++++++++++++++++ sound/soc/codecs/sigmadsp.c | 65 ++------------------------------------ sound/soc/codecs/sigmadsp.h | 20 ++++++++++++ 6 files changed, 107 insertions(+), 65 deletions(-) create mode 100644 sound/soc/codecs/sigmadsp-i2c.c create mode 100644 sound/soc/codecs/sigmadsp-regmap.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cbfa1e1..0b9571c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -225,11 +225,11 @@ config SND_SOC_ADAU1373 config SND_SOC_ADAU1701 tristate "Analog Devices ADAU1701 CODEC" depends on I2C - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_I2C config SND_SOC_ADAU17X1 tristate - select SND_SOC_SIGMADSP + select SND_SOC_SIGMADSP_REGMAP config SND_SOC_ADAU1761 tristate @@ -476,6 +476,14 @@ config SND_SOC_SIGMADSP tristate select CRC32 +config SND_SOC_SIGMADSP_I2C + tristate + select SND_SOC_SIGMADSP + +config SND_SOC_SIGMADSP_REGMAP + tristate + select SND_SOC_SIGMADSP + config SND_SOC_SIRF_AUDIO_CODEC tristate "SiRF SoC internal audio codec" select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index be3377b..1bd6e1c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -77,6 +77,8 @@ snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o +snd-soc-sigmadsp-i2c-objs := sigmadsp-i2c.o +snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-sn95031-objs := sn95031.o @@ -240,6 +242,8 @@ obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o +obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o +obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o diff --git a/sound/soc/codecs/sigmadsp-i2c.c b/sound/soc/codecs/sigmadsp-i2c.c new file mode 100644 index 0000000..246081a --- /dev/null +++ b/sound/soc/codecs/sigmadsp-i2c.c @@ -0,0 +1,35 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/i2c.h> +#include <linux/export.h> +#include <linux/module.h> + +#include "sigmadsp.h" + +static int sigma_action_write_i2c(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return i2c_master_send(control_data, (const unsigned char *)&sa->addr, + len); +} + +int process_sigma_firmware(struct i2c_client *client, const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = client; + ssfw.write = sigma_action_write_i2c; + + return _process_sigma_firmware(&client->dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("SigmaDSP I2C firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp-regmap.c b/sound/soc/codecs/sigmadsp-regmap.c new file mode 100644 index 0000000..f78ed8d --- /dev/null +++ b/sound/soc/codecs/sigmadsp-regmap.c @@ -0,0 +1,36 @@ +/* + * Load Analog Devices SigmaStudio firmware files + * + * Copyright 2009-2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/regmap.h> +#include <linux/export.h> +#include <linux/module.h> + +#include "sigmadsp.h" + +static int sigma_action_write_regmap(void *control_data, + const struct sigma_action *sa, size_t len) +{ + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), + sa->payload, len - 2); +} + +int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, + const char *name) +{ + struct sigma_firmware ssfw; + + ssfw.control_data = regmap; + ssfw.write = sigma_action_write_regmap; + + return _process_sigma_firmware(dev, &ssfw, name); +} +EXPORT_SYMBOL(process_sigma_firmware_regmap); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("SigmaDSP regmap firmware loader"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 4068f24..f2de7e0 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -34,23 +34,6 @@ enum { SIGMA_ACTION_END, }; -struct sigma_action { - u8 instr; - u8 len_hi; - __le16 len; - __be16 addr; - unsigned char payload[]; -} __packed; - -struct sigma_firmware { - const struct firmware *fw; - size_t pos; - - void *control_data; - int (*write)(void *control_data, const struct sigma_action *sa, - size_t len); -}; - static inline u32 sigma_action_len(struct sigma_action *sa) { return (sa->len_hi << 16) | le16_to_cpu(sa->len); @@ -138,7 +121,7 @@ process_sigma_actions(struct sigma_firmware *ssfw) return 0; } -static int _process_sigma_firmware(struct device *dev, +int _process_sigma_firmware(struct device *dev, struct sigma_firmware *ssfw, const char *name) { int ret; @@ -197,50 +180,6 @@ static int _process_sigma_firmware(struct device *dev, return ret; } - -#if IS_ENABLED(CONFIG_I2C) - -static int sigma_action_write_i2c(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return i2c_master_send(control_data, (const unsigned char *)&sa->addr, - len); -} - -int process_sigma_firmware(struct i2c_client *client, const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = client; - ssfw.write = sigma_action_write_i2c; - - return _process_sigma_firmware(&client->dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware); - -#endif - -#if IS_ENABLED(CONFIG_REGMAP) - -static int sigma_action_write_regmap(void *control_data, - const struct sigma_action *sa, size_t len) -{ - return regmap_raw_write(control_data, be16_to_cpu(sa->addr), - sa->payload, len - 2); -} - -int process_sigma_firmware_regmap(struct device *dev, struct regmap *regmap, - const char *name) -{ - struct sigma_firmware ssfw; - - ssfw.control_data = regmap; - ssfw.write = sigma_action_write_regmap; - - return _process_sigma_firmware(dev, &ssfw, name); -} -EXPORT_SYMBOL(process_sigma_firmware_regmap); - -#endif +EXPORT_SYMBOL_GPL(_process_sigma_firmware); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sigmadsp.h b/sound/soc/codecs/sigmadsp.h index e439cbd..c47cd23 100644 --- a/sound/soc/codecs/sigmadsp.h +++ b/sound/soc/codecs/sigmadsp.h @@ -12,6 +12,26 @@ #include <linux/device.h> #include <linux/regmap.h> +struct sigma_action { + u8 instr; + u8 len_hi; + __le16 len; + __be16 addr; + unsigned char payload[]; +} __packed; + +struct sigma_firmware { + const struct firmware *fw; + size_t pos; + + void *control_data; + int (*write)(void *control_data, const struct sigma_action *sa, + size_t len); +}; + +int _process_sigma_firmware(struct device *dev, + struct sigma_firmware *ssfw, const char *name); + struct i2c_client; extern int process_sigma_firmware(struct i2c_client *client, const char *name); -- 1.8.0

2 2

[alsa-devel] [PATCH v2] ASoC: wm5102: Add controls to allow shaping of ultrasonic response
by Charles Keepax 09 Jun '14

09 Jun '14

Add controls to allow custom shaping of the ultrasonic response. This custom shaping can be turned on/off at runtime, although, it should be noted that settings will not affect a currently open audio stream, they will be applied when the next audio stream is started. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Now uses the existing BYTES_EXT control - Delayed other patch in the series, this patch should still work fine, just some pointless caching of the DAC_COMP registers will happen include/linux/mfd/arizona/core.h | 3 ++ sound/soc/codecs/arizona.c | 34 +++++++++++++++++++++ sound/soc/codecs/wm5102.c | 62 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 99 insertions(+), 0 deletions(-) diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index 11783b5..5592651 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -113,6 +113,9 @@ struct arizona { int tdm_width[ARIZONA_MAX_AIF]; int tdm_slots[ARIZONA_MAX_AIF]; + + uint16_t dac_comp_coeff; + uint8_t dac_comp_enabled; }; int arizona_clk32k_enable(struct arizona *arizona); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e77f61c..41b56ee 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1127,6 +1127,31 @@ static int arizona_startup(struct snd_pcm_substream *substream, constraint); } +static void arizona_wm5102_set_dac_comp(struct snd_soc_codec *codec, + unsigned int rate) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + struct reg_default dac_comp[] = { + { 0x80, 0x3 }, + { ARIZONA_DAC_COMP_1, 0 }, + { ARIZONA_DAC_COMP_2, 0 }, + { 0x80, 0x0 }, + }; + + mutex_lock(&codec->mutex); + + dac_comp[1].def = arizona->dac_comp_coeff; + if (rate >= 176400) + dac_comp[2].def = arizona->dac_comp_enabled; + + mutex_unlock(&codec->mutex); + + regmap_multi_reg_write(arizona->regmap, + dac_comp, + ARRAY_SIZE(dac_comp)); +} + static int arizona_hw_params_rate(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -1153,6 +1178,15 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream, switch (dai_priv->clk) { case ARIZONA_CLK_SYSCLK: + switch (priv->arizona->type) { + case WM5102: + arizona_wm5102_set_dac_comp(codec, + params_rate(params)); + break; + default: + break; + } + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, ARIZONA_SAMPLE_RATE_1_MASK, sr_val); if (base) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 289b64d..8613504 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -612,6 +612,62 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + uint16_t data; + + mutex_lock(&codec->mutex); + data = cpu_to_be16(arizona->dac_comp_coeff); + memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + memcpy(&arizona->dac_comp_coeff, ucontrol->value.bytes.data, + sizeof(arizona->dac_comp_coeff)); + arizona->dac_comp_coeff = be16_to_cpu(arizona->dac_comp_coeff); + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = arizona->dac_comp_enabled; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + + mutex_lock(&codec->mutex); + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + mutex_unlock(&codec->mutex); + + return 0; +} + static const char *wm5102_osr_text[] = { "Low power", "Normal", "High performance", }; @@ -843,6 +899,12 @@ SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), SOC_ENUM("Noise Gate Hold", arizona_ng_hold), +SND_SOC_BYTES_EXT("Output Compensation Coefficient", 2, + wm5102_out_comp_coeff_get, wm5102_out_comp_coeff_put), + +SOC_SINGLE_EXT("Output Compensation Switch", 0, 0, 1, 0, + wm5102_out_comp_switch_get, wm5102_out_comp_switch_put), + WM5102_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), WM5102_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), WM5102_NG_SRC("HPOUT2L", ARIZONA_NOISE_GATE_SELECT_2L), -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: rl6231: Remove unneeded inclusion of header files
by Axel Lin 09 Jun '14

09 Jun '14

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rl6231.c | 19 ------------------- 1 file changed, 19 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 7b82fbe..56650d6 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -11,25 +11,6 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/delay.h> -#include <linux/pm.h> -#include <linux/gpio.h> -#include <linux/i2c.h> -#include <linux/regmap.h> -#include <linux/of.h> -#include <linux/of_gpio.h> -#include <linux/platform_device.h> -#include <linux/spi/spi.h> -#include <linux/acpi.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/initval.h> -#include <sound/tlv.h> #include "rl6231.h" -- 1.8.3.2

3 2