November 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Run-time CODEC reset
by Patrick Lai 14 Nov '11

14 Nov '11

Hi, I have a use case which requires CODEC to be reset at run-time. As application exercises playback/capture through PCM device and mixer, my understanding is that there is no interface for application to reset CODEC through ALSA ASoC framework. CODEC driver would have to build non-standard interface for this purpose. Is there precedence in any of asoc CODEC driver which provides reset interface? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

3 3

[alsa-devel] [PATCH 1/1] ASoC: mxs-saif: remove function in platform_data
by Dong Aisheng 14 Nov '11

14 Nov '11

Add master_mode and master_id in platfrom_data since it's board specific and board knows it. Then we can remove the function pointer in platfrom_data to make the driver more devicetree friendly. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> --- This patch is based on Sascha's suggestions and see using in [PATCH v5 0/3] ARM: mxs: add recording support for saif --- include/sound/saif.h | 4 ++-- sound/soc/mxs/mxs-saif.c | 19 ++++++++----------- 2 files changed, 10 insertions(+), 13 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h index d0e0de7..f22f3e1 100644 --- a/include/sound/saif.h +++ b/include/sound/saif.h @@ -10,7 +10,7 @@ #define __SOUND_SAIF_H__ struct mxs_saif_platform_data { - int (*init) (void); - int (*get_master_id) (unsigned int saif_id); + 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 76dc74d..03b64d2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -625,13 +625,6 @@ static int mxs_saif_probe(struct platform_device *pdev) if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; - pdata = pdev->dev.platform_data; - if (pdata && pdata->init) { - ret = pdata->init(); - if (ret) - return ret; - } - saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -639,12 +632,16 @@ static int mxs_saif_probe(struct platform_device *pdev) mxs_saif[pdev->id] = saif; saif->id = pdev->id; - saif->master_id = saif->id; - if (pdata && pdata->get_master_id) { - saif->master_id = pdata->get_master_id(saif->id); + pdata = pdev->dev.platform_data; + if (pdata && !pdata->master_mode) { + saif->master_id = pdata->master_id; if (saif->master_id < 0 || - saif->master_id >= ARRAY_SIZE(mxs_saif)) + saif->master_id >= ARRAY_SIZE(mxs_saif) || + saif->master_id == saif->id) + dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; + } else { + saif->master_id = saif->id; } saif->clk = clk_get(&pdev->dev, NULL); -- 1.7.0.4

4 4

[alsa-devel] snd_pcm_meter_get_rate() (and other meter functions) - can it be used in user applications?
by James Huk 14 Nov '11

14 Nov '11

Hello everybody. I'm trying to understand how to add a sound meter that would monitor sound level on certain channel defined in .asoundrc (like ameter does). I think I should use snd_pcm_meter... however, if I try to use this simple test program: #include <stdio.h> #include <alsa/asoundlib.h> snd_pcm_t *pcm; snd_pcm_scope_t *scope; int main() { int err=snd_pcm_open(&pcm, "meter-x", SND_PCM_STREAM_PLAYBACK,0); snd_pcm_scope_malloc(&scope); snd_pcm_meter_add_scope(pcm, scope); snd_pcm_meter_get_rate(pcm); return 0; } I get: pcm_meter.c:906: snd_pcm_meter_get_rate: Assertion `meter->gen.slave->setup' failed. I don't understand why - is it because snd_pcm_meter functions can only be used in external plugins, and not in user applications (since similar construction works fine with ameter...), or did I missed something? Thanks in advance.

1 0

[alsa-devel] [PATCH] [Trivial] sound, realtek: remove redundant semicolon
by Jesper Juhl 14 Nov '11

14 Nov '11

Having just one semicolon after a break statement is enough. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 308bb57..336d14e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1452,7 +1452,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) switch (fix->type) { case ALC_FIXUP_SKU: if (action != ALC_FIXUP_ACT_PRE_PROBE || !fix->v.sku) - break;; + break; snd_printdd(KERN_INFO "hda_codec: %s: " "Apply sku override for %s\n", codec->chip_name, modelname); -- 1.7.7.3 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

2 1

[alsa-devel] [PATCH] ASoC: davinci: drop SFFSDR_FPGA and CONFIG_SFFSDR_FPGA
by Paul Bolle 13 Nov '11

13 Nov '11

Commit 08bd168696 ("ASoC: Add driver for the Lyrtech SFFSDR board") added a reference to a Kconfig symbol SFFSDR_FPGA. Commit ac37373b64 ("ASoC: DaVinci: Fix SFFSDR compilation error.") added a macro CONFIG_SFFSDR_FPGA. But there's no Kconfig symbol SFFSDR_FPGA and no define of CONFIG_SFFSDR_FPGA. So the select of SFFSDR_FPGA is a nop. It can be dropped. The checks for CONFIG_SFFSDR_FPGA can also be dropped. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) Untested, as this needs (build, etc.) testing if we want to be thorough. But I do not have the hardware and neither do I have the needed toolchain at hand. 1) Note that there's no file named *plat-sffsdr/sffsdr-fpga.h in the tree. Neither is there a function or a macro named sffsdr_fpga_set_codec_fs. So defining CONFIG_SFFSDR_FPGA will never work. sound/soc/davinci/Kconfig | 1 - sound/soc/davinci/davinci-sffsdr.c | 9 --------- 2 files changed, 0 insertions(+), 10 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 9e11a14..6a91206 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -59,7 +59,6 @@ config SND_DAVINCI_SOC_SFFSDR depends on SND_DAVINCI_SOC && MACH_SFFSDR select SND_DAVINCI_SOC_I2S select SND_SOC_PCM3008 - select SFFSDR_FPGA help Say Y if you want to add support for SoC audio on Lyrtech SFFSDR board. diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 0fe558c..d650c2d 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -24,9 +24,6 @@ #include <asm/dma.h> #include <asm/mach-types.h> -#ifdef CONFIG_SFFSDR_FPGA -#include <asm/plat-sffsdr/sffsdr-fpga.h> -#endif #include <mach/edma.h> @@ -53,13 +50,11 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, /* Fsref can be 32000, 44100 or 48000. */ fs = params_rate(params); -#ifndef CONFIG_SFFSDR_FPGA /* Without the FPGA module, the Fs is fixed at 44100 Hz */ if (fs != 44100) { pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); return -EINVAL; } -#endif /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); @@ -68,11 +63,7 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); -#ifndef CONFIG_SFFSDR_FPGA return 0; -#else - return sffsdr_fpga_set_codec_fs(fs); -#endif } static struct snd_soc_ops sffsdr_ops = { -- 1.7.4.4

1 0

Re: [alsa-devel] SPDIF receiver with dynamically changing sample rate
by Clemens Ladisch 13 Nov '11

13 Nov '11

Sam Hawkens wrote: >Some SPDIF receivers have an interrupt output that can be configured to >be asserted when sample rate changes. If the driver supports this, it will notify you of changes of the mixer control that contains the status bits. Regards, Clemens

1 0

Re: [alsa-devel] SPDIF receiver with dynamically changing sample rate
by Sam Hawkens 13 Nov '11

13 Nov '11

Am 12.11.2011 16:36, schrieb Clemens Ladisch: > Most drivers ignore the rate requested by the application and return the samples > as they come from the input. Only the AK411* drivers monitor the rate and stop > the stream on changes. OK, so that should work. >> If yes, how can the application notice the change in sample rate? > This is usually not possible without measuring the rate at which samples arrive. > However, you could check the declared rate of the sender by monitoring the > channel status bits. OK, so this could be a solution as long as the SPDIF stream carries correct channel stauts information (which is not alway the case). Still a good idea. Some SPDIF receivers have an interrupt output that can be configured to be asserted when sample rate changes. With a custom hardware design that's another option in addition to the available ALSA features. Thanx a lot for the good advice! Regards, Sam

1 0

[alsa-devel] SPDIF receiver with dynamically changing sample rate
by Sam Hawkens 12 Nov '11

12 Nov '11

Hi, I want to do the following: write an application that has its input fed from an SPDIF receiver without using sample rate conversion. The sample rate must not be set by the application but must be defined by the SPDIF receiver's PLL. Is it possible to run the application continously with ALSA even if the sample rate changes? If yes, how can the application notice the change in sample rate? Regards Sam

2 1

[PATCH] ALSA: HDA: Jack: Export required functions from hda_jack.c
by David Henningsson 11 Nov '11

11 Nov '11

These two functions are being used by the codec-idt and codec-hdmi modules, so they need to be exported properly. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 1389958..eac002d 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -87,6 +87,7 @@ snd_hda_jack_tbl_new(struct hda_codec *codec, hda_nid_t nid) jack->tag = codec->jacktbl.used; return jack; } +EXPORT_SYMBOL_HDA(snd_hda_jack_tbl_new); #ifdef CONFIG_SND_HDA_INPUT_JACK static void snd_hda_input_jack_free(struct hda_codec *codec); @@ -224,6 +225,7 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->kctl = kctl; return 0; } +EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, int idx, const struct auto_pin_cfg *cfg) -- 1.7.5.4 --------------030403010104080501020808 Content-Type: text/plain; name="dmesg.txt" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename="dmesg.txt" [ 17.249226] snd_hda_intel 0000:00:1b.0: power state changed by ACPI to D0 [ 17.249239] snd_hda_intel 0000:00:1b.0: power state changed by ACPI to D0 [ 17.249252] snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 [ 17.249353] snd_hda_intel 0000:00:1b.0: irq 47 for MSI/MSI-X [ 17.249393] snd_hda_intel 0000:00:1b.0: setting latency timer to 64 [ 17.249400] ALSA hda_intel.c:2775 chipset global capabilities = 0x4401 [ 17.249418] ALSA hda_intel.c:1175 Clearing TCSEL [ 17.278073] ALSA hda_intel.c:1021 codec_mask = 0x9 [ 17.281330] ALSA hda_intel.c:1484 codec #0 probed OK [ 17.281396] ALSA hda_intel.c:1484 codec #3 probed OK [ 17.302729] ALSA patch_sigmatel.c:5622 hda_codec: 92HD81B1X5: BIOS auto-probing. [ 17.302906] ALSA patch_sigmatel.c:5635 mute LED gpio 1 polarity 0 [ 17.302914] ALSA hda_codec.c:4917 autoconfig: line_outs=1 (0xd/0x0/0x0/0x0/0x0) type:speaker [ 17.302918] ALSA hda_codec.c:4921 speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 17.302922] ALSA hda_codec.c:4925 hp_outs=1 (0xb/0x0/0x0/0x0/0x0) [ 17.302925] ALSA hda_codec.c:4926 mono: mono_out=0x0 [ 17.302927] ALSA hda_codec.c:4930 inputs: [ 17.302931] ALSA hda_codec.c:4934 Mic=0xa [ 17.302933] ALSA hda_codec.c:4936 [ 17.303160] ALSA patch_sigmatel.c:3123 stac92xx: dac_nids=1 (0x13/0x0/0x0/0x0/0x0) [ 17.303392] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input6 [ 17.303918] Linux video capture interface: v2.00 [ 17.304759] uvcvideo: Found UVC 1.00 device HP Webcam [2 MP Fixed] (04f2:b1ac) [ 17.312696] ALSA patch_sigmatel.c:4135 stac_gpio_set msk 1 dir 1 gpio 1 [ 17.318851] BUG: unable to handle kernel NULL pointer dereference at 0000000000000078 [ 17.318917] IP: [<ffffffffa018544c>] snd_kctl_jack_report+0xc/0x30 [snd] [ 17.318976] PGD a2a8d067 PUD a2a8c067 PMD 0 [ 17.319013] Oops: 0000 [#1] SMP [ 17.319041] CPU 3 [ 17.319056] Modules linked in: snd_hda_codec_hdmi uvcvideo(+) videodev snd_hda_codec_idt v4l2_compat_ioctl32 snd_hda_intel(+) snd_hda_codec snd_hwdep snd_pcm snd_rawmidi binfmt_misc snd_seq_midi_event snd_seq psmouse serio_raw intel_ips arc4 hp_accel lis3lv02d snd_seq_device snd_timer input_polldev ath9k mac80211 ath9k_common ath9k_hw ath snd cfg80211 soundcore snd_page_alloc mei(C) lp parport wmi i915 r8169 drm_kms_helper ahci drm libahci i2c_algo_bit video [ 17.319399] [ 17.319412] Pid: 475, comm: modprobe Tainted: G C 3.0.0-12-generic #20-Ubuntu Hewlett-Packard HP ProBook 4520s/1413 [ 17.319492] RIP: 0010:[<ffffffffa018544c>] [<ffffffffa018544c>] snd_kctl_jack_report+0xc/0x30 [snd] [ 17.319561] RSP: 0018:ffff8800afa8bb88 EFLAGS: 00010246 [ 17.319599] RAX: 0000000000000000 RBX: ffff8800b1b35828 RCX: ffff8800b17b6428 [ 17.319649] RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffff880035e4e428 [ 17.319699] RBP: ffff8800afa8bb88 R08: 0000000000000020 R09: 0000000000000000 [ 17.319748] R10: 0000000000000001 R11: 0000000000000000 R12: 0000000000000000 [ 17.319797] R13: ffff8800a0a70028 R14: 000000000000000a R15: 0000000000000001 [ 17.319848] FS: 00007fd24a8b1720(0000) GS:ffff8800b6cc0000(0000) knlGS:0000000000000000 [ 17.319904] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 17.319944] CR2: 0000000000000078 CR3: 00000000afad0000 CR4: 00000000000006e0 [ 17.319993] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 17.320042] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 17.320092] Process modprobe (pid: 475, threadinfo ffff8800afa8a000, task ffff880035e0dc80) [ 17.320148] Stack: [ 17.320163] ffff8800afa8bbb8 ffffffffa02e7208 000000000000000a ffff8800a2be2028 [ 17.320219] 0000000000000001 ffff8800a0a70028 ffff8800afa8bc08 ffffffffa0316fee [ 17.320275] ffff8800a0a70828 0000000000000002 ffff8800afa8bc38 ffff8800a0a70028 [ 17.320331] Call Trace: [ 17.320358] [<ffffffffa02e7208>] snd_hda_jack_report_sync+0x48/0x90 [snd_hda_codec] [ 17.320417] [<ffffffffa0316fee>] stac92xx_init+0x6be/0x820 [snd_hda_codec_idt] [ 17.320472] [<ffffffffa02e69f0>] snd_hda_codec_build_controls+0x30/0x60 [snd_hda_codec] [ 17.320532] [<ffffffffa02e6a58>] snd_hda_build_controls+0x38/0x90 [snd_hda_codec] [ 17.320589] [<ffffffffa02bda93>] azx_probe+0x99c/0xa36 [snd_hda_intel] [ 17.320639] [<ffffffff8130f32c>] local_pci_probe+0x5c/0xd0 [ 17.320680] [<ffffffff81310b39>] __pci_device_probe+0xf9/0x100 [ 17.320723] [<ffffffff812e9c3a>] ? kobject_get+0x1a/0x30 [ 17.320761] [<ffffffff81310b7a>] pci_device_probe+0x3a/0x60 [ 17.320805] [<ffffffff813ce4e8>] really_probe+0x68/0x190 [ 17.320843] [<ffffffff813ce775>] driver_probe_device+0x45/0x70 [ 17.320885] [<ffffffff813ce84b>] __driver_attach+0xab/0xb0 [ 17.320925] [<ffffffff813ce7a0>] ? driver_probe_device+0x70/0x70 [ 17.320970] [<ffffffff813ce7a0>] ? driver_probe_device+0x70/0x70 [ 17.321014] [<ffffffff813cd5dc>] bus_for_each_dev+0x5c/0x90 [ 17.321054] [<ffffffff813ce2ae>] driver_attach+0x1e/0x20 [ 17.321093] [<ffffffff813cdf00>] bus_add_driver+0x1a0/0x270 [ 17.321135] [<ffffffffa030d000>] ? 0xffffffffa030cfff [ 17.321172] [<ffffffff813cedb6>] driver_register+0x76/0x140 [ 17.321212] [<ffffffffa030d000>] ? 0xffffffffa030cfff [ 17.323078] [<ffffffff81310816>] __pci_register_driver+0x56/0xd0 [ 17.324999] [<ffffffff8103b103>] ? set_memory_nx+0x43/0x50 [ 17.326918] [<ffffffffa030d01e>] alsa_card_azx_init+0x1e/0x1000 [snd_hda_intel] [ 17.326927] [<ffffffff81002042>] do_one_initcall+0x42/0x180 [ 17.326931] [<ffffffff8109ff2e>] sys_init_module+0xbe/0x230 [ 17.326937] [<ffffffff815f22c2>] system_call_fastpath+0x16/0x1b [ 17.326939] Code: 89 63 4c 48 c7 43 78 00 00 00 00 48 89 d8 4c 8b 65 f0 48 8b 5d e8 4c 8b 6d f8 c9 c3 0f 1f 00 55 48 89 e5 66 66 66 66 90 0f b6 d2 [ 17.326955] 39 56 78 74 12 48 89 56 78 48 8d 56 10 be 01 00 00 00 e8 bc [ 17.326964] RIP [<ffffffffa018544c>] snd_kctl_jack_report+0xc/0x30 [snd] [ 17.326977] RSP <ffff8800afa8bb88> [ 17.326979] CR2: 0000000000000078 [ 17.344773] init: failsafe main process (759) killed by TERM signal --------------030403010104080501020808--

1 0

[alsa-devel] [PATCH v2] ARM: pxa: add support for Raumfeld DDX
by Daniel Mack 11 Nov '11

11 Nov '11

This new product features a STA32x codec. To support it, some rework of the regulator initialization for the Raumfeld platform is necessary. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Mark Brown <broonie at opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg at ti.com> Cc: Eric Miao <eric.y.miao at gmail.com> Cc: Sven Neumann <s.neumann(a)raumfeld.com> Cc: Johannes Stezenbach <js(a)sig21.net> --- v2 of this patch has clocking fixes and superseeds the older version. As there are more changes in sound/soc than in arch/arm, I think it should best go through Mark's AsoC tree. arch/arm/mach-pxa/raumfeld.c | 119 +++++++++++++++++--- sound/soc/pxa/Kconfig | 2 + sound/soc/pxa/raumfeld.c | 247 +++++++++++++++++++++++++++++++++++++++--- 3 files changed, 336 insertions(+), 32 deletions(-) diff --git a/arch/arm/mach-pxa/raumfeld.c b/arch/arm/mach-pxa/raumfeld.c index 7856fe4..22366ff 100644 --- a/arch/arm/mach-pxa/raumfeld.c +++ b/arch/arm/mach-pxa/raumfeld.c @@ -850,7 +850,7 @@ struct regulator_init_data audio_va_initdata = { }, }; -static struct fixed_voltage_config audio_va_config = { +static struct fixed_voltage_config cs4270_va_config = { .supply_name = "audio_va", .microvolts = 5000000, .gpio = GPIO_AUDIO_VA_ENABLE, @@ -859,47 +859,114 @@ static struct fixed_voltage_config audio_va_config = { .init_data = &audio_va_initdata, }; -static struct platform_device audio_va_device = { +static struct platform_device cs4270_va_device = { .name = "reg-fixed-voltage", .id = 0, .dev = { - .platform_data = &audio_va_config, + .platform_data = &cs4270_va_config, }, }; /* Dummy supplies for Codec's VD/VLC */ -static struct regulator_consumer_supply audio_dummy_supplies[] = { +static struct regulator_consumer_supply cs4270_dummy_supplies[] = { REGULATOR_SUPPLY("vd", "0-0048"), REGULATOR_SUPPLY("vlc", "0-0048"), }; -struct regulator_init_data audio_dummy_initdata = { - .consumer_supplies = audio_dummy_supplies, - .num_consumer_supplies = ARRAY_SIZE(audio_dummy_supplies), +struct regulator_init_data cs4270_dummy_initdata = { + .consumer_supplies = cs4270_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(cs4270_dummy_supplies), .constraints = { .valid_ops_mask = REGULATOR_CHANGE_STATUS, }, }; -static struct fixed_voltage_config audio_dummy_config = { - .supply_name = "audio_vd", +static struct fixed_voltage_config cs4270_dummy_config = { + .supply_name = "cs4270_vd", .microvolts = 3300000, .gpio = -1, - .init_data = &audio_dummy_initdata, + .init_data = &cs4270_dummy_initdata, }; -static struct platform_device audio_supply_dummy_device = { +static struct platform_device cs4270_supply_dummy_device = { .name = "reg-fixed-voltage", .id = 1, .dev = { - .platform_data = &audio_dummy_config, + .platform_data = &cs4270_dummy_config, }, }; -static struct platform_device *audio_regulator_devices[] = { - &audio_va_device, - &audio_supply_dummy_device, +static struct platform_device *cs4270_regulator_devices[] = { + &cs4270_va_device, + &cs4270_supply_dummy_device, +}; + +/* Fixed regulator for sta32x Vdda supply + * 0-001a maps to the sta32x codec devname (derived from i2c bus num + addr) + */ + +static struct regulator_consumer_supply sta32x_va_consumer_supply = + REGULATOR_SUPPLY("Vdda", "0-001a"); + +struct regulator_init_data sta32x_va_initdata = { + .consumer_supplies = &sta32x_va_consumer_supply, + .num_consumer_supplies = 1, + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_va_config = { + .supply_name = "audio_va", + .microvolts = 3300000, + .gpio = GPIO_AUDIO_VA_ENABLE, + .enable_high = 1, + .enabled_at_boot = 0, + .init_data = &sta32x_va_initdata, +}; + +static struct platform_device sta32x_va_device = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &sta32x_va_config, + }, +}; + +/* Dummy supplies for sta32x codec's Vdd3/Vcc */ + +static struct regulator_consumer_supply sta32x_dummy_supplies[] = { + REGULATOR_SUPPLY("Vdd3", "0-001a"), + REGULATOR_SUPPLY("Vcc", "0-001a"), +}; + +struct regulator_init_data sta32x_dummy_initdata = { + .consumer_supplies = sta32x_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(sta32x_dummy_supplies), + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config sta32x_dummy_config = { + .supply_name = "audio_vd", + .microvolts = 3300000, + .gpio = -1, + .init_data = &sta32x_dummy_initdata, +}; + +static struct platform_device sta32x_supply_dummy_device = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &sta32x_dummy_config, + }, +}; + +static struct platform_device *sta32x_regulator_devices[] = { + &sta32x_va_device, + &sta32x_supply_dummy_device, }; /** @@ -948,6 +1015,11 @@ static struct i2c_board_info raumfeld_connector_i2c_board_info __initdata = { .addr = 0x48, }; +static struct i2c_board_info raumfeld_ddx_i2c_board_info __initdata = { + .type = "sta326", + .addr = 0x1a, +}; + static struct eeti_ts_platform_data eeti_ts_pdata = { .irq_active_high = 1, }; @@ -987,7 +1059,10 @@ static void __init raumfeld_audio_init(void) else gpio_direction_output(GPIO_MCLK_RESET, 1); - platform_add_devices(ARRAY_AND_SIZE(audio_regulator_devices)); + if ((system_rev & 0xff00) == 0x0400) + platform_add_devices(ARRAY_AND_SIZE(sta32x_regulator_devices)); + else + platform_add_devices(ARRAY_AND_SIZE(cs4270_regulator_devices)); } static void __init raumfeld_common_init(void) @@ -1060,7 +1135,11 @@ static void __init raumfeld_connector_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_connector_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(connector_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); platform_device_register(&smc91x_device); @@ -1072,7 +1151,11 @@ static void __init raumfeld_speaker_init(void) { pxa3xx_mfp_config(ARRAY_AND_SIZE(raumfeld_speaker_pin_config)); spi_register_board_info(ARRAY_AND_SIZE(speaker_spi_devices)); - i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); + + if ((system_rev & 0xff00) == 0x0400) + i2c_register_board_info(0, &raumfeld_ddx_i2c_board_info, 1); + else + i2c_register_board_info(0, &raumfeld_connector_i2c_board_info, 1); platform_device_register(&smc91x_device); platform_device_register(&rotary_encoder_device); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46..e6db56a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -152,6 +152,8 @@ config SND_SOC_RAUMFELD select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 + select SND_SOC_STA32X + select SND_SOC_WM8782 help Say Y if you want to add support for SoC audio on Raumfeld devices diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 1a591f1..7d4f1bf 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/gpio.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <asm/mach-types.h> @@ -225,6 +226,175 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .hw_params = raumfeld_ak4104_hw_params, }; +/* STA32X */ +static int raumfeld_sta32x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* fixed MCLK of 11.2896MHz */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + + /* PXA DMA cannot do zero extend for 24bit samples, + * thus only 16bit (two samples packet into 32bit word) + * or 32bit samples are possible + */ + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE); + + /* we have a fixed MCLK, set it here so ALSA knows + * the supported sample rates and can resample if necessary + */ + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static void raumfeld_sta32x_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + /* set freq to 0 to enable all possible codec sample rates */ + snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); +} + +static int raumfeld_sta32x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // fixed MCLK of 11.2896MHz + switch (params_rate(params)) { + case 44100: + case 88200: + case 176400: + clk = 11289600; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + /* this enables network mode for 2 * 32bit samples */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 0, 2, 32); + if (ret < 0) + return ret; + break; + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + /* this disables network mode */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0, 0, 0, 16); + if (ret < 0) + return ret; + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_sta32x_ops = { + .startup = raumfeld_sta32x_startup, + .shutdown = raumfeld_sta32x_shutdown, + .hw_params = raumfeld_sta32x_hw_params, +}; + +/* WM8782 */ +static int raumfeld_wm8782_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + // fixed MCLK of 11.2896MHz + switch (params_rate(params)) { + case 44100: + clk = 11289600; + break; + default: + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + break; + default: + return -EINVAL; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_wm8782_ops = { + .hw_params = raumfeld_wm8782_hw_params, +}; + #define DAI_LINK_CS4270 \ { \ .name = "CS4270", \ @@ -247,6 +417,28 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .codec_name = "spi0.0", \ } +#define DAI_LINK_STA32X \ +{ \ + .name = "STA32X", \ + .stream_name = "Playback", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "STA32X", \ + .ops = &raumfeld_sta32x_ops, \ + .codec_name = "sta32x.0-001a", \ +} + +#define DAI_LINK_WM8782 \ +{ \ + .name = "wm8782", \ + .stream_name = "wm8782", \ + .cpu_dai_name = "pxa-ssp-dai.0", \ + .platform_name = "pxa-pcm-audio", \ + .codec_dai_name = "wm8782", \ + .codec_name = "wm8782.0", \ + .ops = &raumfeld_wm8782_ops, \ +} + static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] = { DAI_LINK_CS4270, @@ -258,6 +450,12 @@ static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] = DAI_LINK_CS4270, }; +static struct snd_soc_dai_link snd_soc_raumfeld_ddx_dai[] = +{ + DAI_LINK_STA32X, + DAI_LINK_WM8782, +}; + static struct snd_soc_card snd_soc_raumfeld_connector = { .name = "Raumfeld Connector", .dai_link = snd_soc_raumfeld_connector_dai, @@ -274,7 +472,15 @@ static struct snd_soc_card snd_soc_raumfeld_speaker = { .resume_pre = raumfeld_analog_resume, }; -static struct platform_device *raumfeld_audio_device; +static struct snd_soc_card snd_soc_raumfeld_ddx = { + .name = "Raumfeld DDX", + .dai_link = snd_soc_raumfeld_ddx_dai, + .num_links = ARRAY_SIZE(snd_soc_raumfeld_ddx_dai), + .suspend_post = raumfeld_analog_suspend, + .resume_pre = raumfeld_analog_resume, +}; + +static struct platform_device *raumfeld_audio_device, *wm8782_device; static int __init raumfeld_audio_init(void) { @@ -284,26 +490,39 @@ static int __init raumfeld_audio_init(void) !machine_is_raumfeld_connector()) return 0; - max9486_client = i2c_new_device(i2c_get_adapter(0), - &max9486_hwmon_info); + if ((system_rev & 0xff00) == 0x0400) { + wm8782_device = platform_device_alloc("wm8782", 0); + if (!wm8782_device) + return -ENOMEM; - if (!max9486_client) - return -ENOMEM; + platform_device_add(wm8782_device); + } else { + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; - set_max9485_clk(MAX9485_MCLK_FREQ_122880); + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + } - /* Register analog device */ + /* Register audio device */ raumfeld_audio_device = platform_device_alloc("soc-audio", 0); if (!raumfeld_audio_device) return -ENOMEM; - if (machine_is_raumfeld_speaker()) + if ((system_rev & 0xff00) == 0x0400) platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_speaker); - - if (machine_is_raumfeld_connector()) - platform_set_drvdata(raumfeld_audio_device, - &snd_soc_raumfeld_connector); + &snd_soc_raumfeld_ddx); + else { + if (machine_is_raumfeld_speaker()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_speaker); + + if (machine_is_raumfeld_connector()) + platform_set_drvdata(raumfeld_audio_device, + &snd_soc_raumfeld_connector); + } ret = platform_device_add(raumfeld_audio_device); if (ret < 0) @@ -317,8 +536,8 @@ static void __exit raumfeld_audio_exit(void) { raumfeld_enable_audio(false); + platform_device_unregister(wm8782_device); platform_device_unregister(raumfeld_audio_device); - i2c_unregister_device(max9486_client); gpio_free(GPIO_MCLK_RESET); -- 1.7.6.4

1 1