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[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--

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[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

[alsa-devel] Problem with bluetooth headset and "aplay -l"
by Anders Gnistrup 11 Nov '11

11 Nov '11

Hi I have a wired problem. I can pair, and connect a bluetooth headset using bluez-simple-agent and bluez-test-audio, no problem. I kan also record and play using aplay/arecord -D bluetooth. But i can list the device in "aplay -l" og "aplay -L". My gees is that I am missing a ctl interface or a kernel module. (or that ubuntu has done something nasty) My system is a ubuntu 11.04, where the bluetooth pulseaudion plugin have been removed. I am developing for a embedded arm platform also running ubunut 11.04 and the problem is the same on target. The list of modules is listed in the bottom og this email. Regards Anders Gnistrup Module Size Used by aesni_intel 55161 1 cryptd 20510 1 aesni_intel aes_x86_64 17208 1 aesni_intel aes_generic 38279 2 aesni_intel,aes_x86_64 parport_pc 36959 0 ppdev 17113 0 snd_hda_codec_hdmi 28167 1 snd_hda_codec_idt 71137 1 binfmt_misc 17565 1 pata_pcmcia 17074 1 snd_hda_intel 33176 2 snd_hda_codec 103804 3 snd_hda_codec_hdmi,snd_hda_codec_idt,snd_hda_intel snd_hwdep 13604 1 snd_hda_codec rfcomm 47694 19 snd_pcm 96391 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_seq_midi 13324 0 sco 18187 2 arc4 12529 2 snd_rawmidi 30486 1 snd_seq_midi bnep 18308 2 snd_seq_midi_event 14899 1 snd_seq_midi l2cap 53610 28 rfcomm,bnep snd_seq 61621 2 snd_seq_midi,snd_seq_midi_event tpm_infineon 17572 0 iwlagn 333716 0 i915 515078 8 snd_timer 29602 2 snd_pcm,snd_seq drm_kms_helper 42394 1 i915 iwlcore 167502 1 iwlagn hp_wmi 13706 0 pcmcia 49166 1 pata_pcmcia joydev 17606 0 sparse_keymap 13898 1 hp_wmi snd_seq_device 14462 3 snd_seq_midi,snd_rawmidi,snd_seq mac80211 294370 2 iwlagn,iwlcore drm 227534 4 i915,drm_kms_helper yenta_socket 27846 0 psmouse 73535 0 pcmcia_rsrc 18372 1 yenta_socket cfg80211 178528 3 iwlagn,iwlcore,mac80211 snd 67382 14 snd_hda_codec_hdmi,snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device tpm_tis 18537 0 btusb 18600 6 bluetooth 72320 11 rfcomm,sco,bnep,l2cap,btusb soundcore 12680 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm hp_accel 21880 0 serio_raw 13166 0 tpm 22267 2 tpm_infineon,tpm_tis lis3lv02d 19893 1 hp_accel tpm_bios 13684 1 tpm lp 17825 0 input_polldev 14007 1 lis3lv02d parport 46458 3 parport_pc,ppdev,lp pcmcia_core 22569 3 pcmcia,yenta_socket,pcmcia_rsrc intel_ips 18097 0 i2c_algo_bit 13400 1 i915 video 19438 1 i915 hid_microsoft 12865 0 usb_storage 53538 0 uas 17996 0 hid_logitech 17693 0 ff_memless 13097 1 hid_logitech usbhid 46956 1 hid_logitech hid 91020 3 hid_microsoft,hid_logitech,usbhid firewire_ohci 40370 0 sdhci_pci 13989 0 firewire_core 62646 1 firewire_ohci sdhci 27387 1 sdhci_pci crc_itu_t 12707 1 firewire_core ahci 25951 2 libahci 26642 1 ahci e1000e 159096 0

2 2

[alsa-devel] Need help with "BUG: key e69d3aac not in .data!"
by Timur Tabi 11 Nov '11

11 Nov '11

I turned on some mutex debugging options, and now I'm getting this in my audio driver: Freescale Synchronous Serial Interface (SSI) ASoC Driver BUG: key e69d3aac not in .data! ------------[ cut here ]------------ WARNING: at /home/b04825/git/linux.10b/kernel/lockdep.c:2980 Modules linked in: NIP: c0078160 LR: c0078148 CTR: 00000000 REGS: e685dcd0 TRAP: 0700 Not tainted (3.2.0-10b-00093-gebea711-dirty) MSR: 00029000 <EE,ME,CE> CR: 22044042 XER: 20000000 TASK = e6860000[1] 'swapper' THREAD: e685c000 CPU: 0 GPR00: 00000000 e685dd80 e6860000 00000001 00000001 c0043824 00000000 00000002 GPR08: 00000000 c0620000 00000001 00000000 22044042 40401800 00000000 00000000 GPR16: c0000a00 00000014 3fffffff 03ff9000 00000015 7ff3a68c c061e000 00000000 GPR24: 00000000 00000000 00001700 fffffff4 00000124 e6f9a3c0 c0620000 e6f9a3c8 NIP [c0078160] lockdep_init_map+0x25c/0x468 LR [c0078148] lockdep_init_map+0x244/0x468 Call Trace: [e685dd80] [c0078138] lockdep_init_map+0x234/0x468 (unreliable) [e685ddc0] [c0133740] sysfs_add_file_mode+0x98/0xf8 [e685ddf0] [c0277578] device_create_file+0x20/0x30 [e685de00] [c0462754] fsl_ssi_probe+0x284/0x410 [e685deb0] [c027ba2c] platform_drv_probe+0x20/0x30 [e685dec0] [c027a4b0] driver_probe_device+0xa4/0x1d4 [e685dee0] [c027a6a4] __driver_attach+0xc4/0xc8 [e685df00] [c027939c] bus_for_each_dev+0x60/0x9c [e685df30] [c027a0e4] driver_attach+0x24/0x34 [e685df40] [c0279d30] bus_add_driver+0x1b0/0x278 [e685df70] [c027aab8] driver_register+0x88/0x154 [e685df90] [c027bd5c] platform_driver_register+0x68/0x78 [e685dfa0] [c05d822c] fsl_ssi_init+0x24/0x34 [e685dfb0] [c0001eb8] do_one_initcall+0x34/0x1ac [e685dfe0] [c05b984c] kernel_init+0xa0/0x13c [e685dff0] [c000e588] kernel_thread+0x4c/0x68 Instruction dump: 7ca42b78 483e03b9 801e5158 2f800000 409efe30 4819def5 2f830000 41befe24 3d20c062 80095328 2f800000 40befe14 <0fe00000> 4bfffe0c 3f9c000a 881cc010 ---[ end trace d5a900576f6c55ce ]--- Is there something wrong with the way I'm calling device_create_file()? That code has been around for years. I don't understand what the kernel is complainging about. -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] [PATCH 1/2] ASoC: Remove unused defines in alc5632 codec.
by Leon Romanovsky 11 Nov '11

11 Nov '11

Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 21 --------------------- 1 files changed, 0 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 68465d8..0fff98e 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -964,27 +964,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) return 0; } -#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ - | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ - | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ - | ALC5632_ADC_REC_MONOMIX) - -#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ - | ALC5632_MIC_ROUTE_SPK \ - | ALC5632_MIC_ROUTE_MONOMIX) - -#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ - | ALC5632_PWR_DAC_STATUS \ - | ALC5632_PWR_AMIX_STATUS \ - | ALC5632_PWR_VREF_STATUS) - -#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ - / ALC5632_ADC_REC_GAIN_STEP) - -#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) - -#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) - static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); -- 1.7.3.4

2 2

[alsa-devel] question to some webcam issue
by Alexey Fisher 11 Nov '11

11 Nov '11

Hi all, i working on some issues of usb webcams, especially my current priority is uvcvideo driver. with some previous kernel version i expoirianced some really nasty crashes if i used skype. i think the problem was the control short timeout for control traffic. i think it was encreesed now to 1000 again. on one side i do not have this problem any more, but on other side i think it should be prewented for future. Description of problem (the way i think it is): - skype usually use really intensively volume controls for echo cancellation and proper record level. - if webcam sound is used then video traffic encreese control delays for audio. - if usb-audio sees this big delay, it will reset the usb device ... complete, even video part. - in this situation many part will die... uvcvideo will be reseted and need to cut access to /dev/video0 the capture programm will freeze for ever. pulseaudio freeze as well and some time you need to restart computer to bring it all back. i know, other parts should be more error resistant, but may be we can prevent at least a part of it. For example do some sort of usb lock if one device has two different interfaces and two of them used, one should not kill another one? Hope you understand my description, Regards, Alexey.

1 0

[alsa-devel] [PATCH v4 0/3] ARM: mxs: add recording support for saif
by Dong Aisheng 11 Nov '11

11 Nov '11

Changes since v3: * remove unneeded locking according to Sascha. The patches are based on imx-features branch since commit: ca4e419c2. Changes since v2: * separate clkmux code into another patch according to Uwe * add lock according to Wolfram * Other minus fixes suggested by Uwe and Wolfram. * remove Wolfram's fix saif clock setting patch which is in v2 series since Wolfram will reform the clock code and send it out himself. For test purpose, user still needs that patch. People can get it from v2 series. The patch is: [PATCH v2 3/3] arm: mxs: disable clock-gates when setting saif-clocks The patches are based on imx-features branch since commit: f4f01e31835f. Changes since v1: The main changes are move mach-specific code(clkmux in DIGCTL) from saif driver to mach-specific layer based on Wolfram's suggestion. Note that the last patch is a RFC patch and sent out for testing since without that patch the saif may not work. Dong Aisheng (3): ARM: mxs: add saif clkmux functions ARM: mx28evk: add platform data for saif ARM: mx28evk: set a initial clock rate for saif arch/arm/mach-mxs/clock-mx28.c | 65 +++++++++++++++++++++++ arch/arm/mach-mxs/devices-mx28.h | 3 +- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 +- arch/arm/mach-mxs/include/mach/common.h | 2 + arch/arm/mach-mxs/include/mach/devices-common.h | 4 +- arch/arm/mach-mxs/include/mach/digctl.h | 21 +++++++ arch/arm/mach-mxs/mach-mx28evk.c | 16 +++++- 7 files changed, 110 insertions(+), 6 deletions(-) create mode 100644 arch/arm/mach-mxs/include/mach/digctl.h

4 12

[alsa-devel] [PATCH] ASoC: Convert WM8995 to direct regcache usage
by Mark Brown 10 Nov '11

10 Nov '11

Large code size increase due to the addition of readability information and the reformatting of the defaults table. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8995.c | 717 ++++++++++++++++++++++++++++++++++++++------- 1 files changed, 608 insertions(+), 109 deletions(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4d109b1..3774acb 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/spi/spi.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> @@ -43,88 +44,331 @@ static const char *wm8995_supply_names[WM8995_NUM_SUPPLIES] = { "MICVDD" }; -static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = { - [0] = 0x8995, [5] = 0x0100, [16] = 0x000b, [17] = 0x000b, - [24] = 0x02c0, [25] = 0x02c0, [26] = 0x02c0, [27] = 0x02c0, - [28] = 0x000f, [32] = 0x0005, [33] = 0x0005, [40] = 0x0003, - [41] = 0x0013, [48] = 0x0004, [56] = 0x09f8, [64] = 0x1f25, - [69] = 0x0004, [82] = 0xaaaa, [84] = 0x2a2a, [146] = 0x0060, - [256] = 0x0002, [257] = 0x8004, [520] = 0x0010, [528] = 0x0083, - [529] = 0x0083, [548] = 0x0c80, [580] = 0x0c80, [768] = 0x4050, - [769] = 0x4000, [771] = 0x0040, [772] = 0x0040, [773] = 0x0040, - [774] = 0x0004, [775] = 0x0100, [784] = 0x4050, [785] = 0x4000, - [787] = 0x0040, [788] = 0x0040, [789] = 0x0040, [1024] = 0x00c0, - [1025] = 0x00c0, [1026] = 0x00c0, [1027] = 0x00c0, [1028] = 0x00c0, - [1029] = 0x00c0, [1030] = 0x00c0, [1031] = 0x00c0, [1056] = 0x0200, - [1057] = 0x0010, [1058] = 0x0200, [1059] = 0x0010, [1088] = 0x0098, - [1089] = 0x0845, [1104] = 0x0098, [1105] = 0x0845, [1152] = 0x6318, - [1153] = 0x6300, [1154] = 0x0fca, [1155] = 0x0400, [1156] = 0x00d8, - [1157] = 0x1eb5, [1158] = 0xf145, [1159] = 0x0b75, [1160] = 0x01c5, - [1161] = 0x1c58, [1162] = 0xf373, [1163] = 0x0a54, [1164] = 0x0558, - [1165] = 0x168e, [1166] = 0xf829, [1167] = 0x07ad, [1168] = 0x1103, - [1169] = 0x0564, [1170] = 0x0559, [1171] = 0x4000, [1184] = 0x6318, - [1185] = 0x6300, [1186] = 0x0fca, [1187] = 0x0400, [1188] = 0x00d8, - [1189] = 0x1eb5, [1190] = 0xf145, [1191] = 0x0b75, [1192] = 0x01c5, - [1193] = 0x1c58, [1194] = 0xf373, [1195] = 0x0a54, [1196] = 0x0558, - [1197] = 0x168e, [1198] = 0xf829, [1199] = 0x07ad, [1200] = 0x1103, - [1201] = 0x0564, [1202] = 0x0559, [1203] = 0x4000, [1280] = 0x00c0, - [1281] = 0x00c0, [1282] = 0x00c0, [1283] = 0x00c0, [1312] = 0x0200, - [1313] = 0x0010, [1344] = 0x0098, [1345] = 0x0845, [1408] = 0x6318, - [1409] = 0x6300, [1410] = 0x0fca, [1411] = 0x0400, [1412] = 0x00d8, - [1413] = 0x1eb5, [1414] = 0xf145, [1415] = 0x0b75, [1416] = 0x01c5, - [1417] = 0x1c58, [1418] = 0xf373, [1419] = 0x0a54, [1420] = 0x0558, - [1421] = 0x168e, [1422] = 0xf829, [1423] = 0x07ad, [1424] = 0x1103, - [1425] = 0x0564, [1426] = 0x0559, [1427] = 0x4000, [1568] = 0x0002, - [1792] = 0xa100, [1793] = 0xa101, [1794] = 0xa101, [1795] = 0xa101, - [1796] = 0xa101, [1797] = 0xa101, [1798] = 0xa101, [1799] = 0xa101, - [1800] = 0xa101, [1801] = 0xa101, [1802] = 0xa101, [1803] = 0xa101, - [1804] = 0xa101, [1805] = 0xa101, [1825] = 0x0055, [1848] = 0x3fff, - [1849] = 0x1fff, [2049] = 0x0001, [2050] = 0x0069, [2056] = 0x0002, - [2057] = 0x0003, [2058] = 0x0069, [12288] = 0x0001, [12289] = 0x0001, - [12291] = 0x0006, [12292] = 0x0040, [12293] = 0x0001, [12294] = 0x000f, - [12295] = 0x0006, [12296] = 0x0001, [12297] = 0x0003, [12298] = 0x0104, - [12300] = 0x0060, [12301] = 0x0011, [12302] = 0x0401, [12304] = 0x0050, - [12305] = 0x0003, [12306] = 0x0100, [12308] = 0x0051, [12309] = 0x0003, - [12310] = 0x0104, [12311] = 0x000a, [12312] = 0x0060, [12313] = 0x003b, - [12314] = 0x0502, [12315] = 0x0100, [12316] = 0x2fff, [12320] = 0x2fff, - [12324] = 0x2fff, [12328] = 0x2fff, [12332] = 0x2fff, [12336] = 0x2fff, - [12340] = 0x2fff, [12344] = 0x2fff, [12348] = 0x2fff, [12352] = 0x0001, - [12353] = 0x0001, [12355] = 0x0006, [12356] = 0x0040, [12357] = 0x0001, - [12358] = 0x000f, [12359] = 0x0006, [12360] = 0x0001, [12361] = 0x0003, - [12362] = 0x0104, [12364] = 0x0060, [12365] = 0x0011, [12366] = 0x0401, - [12368] = 0x0050, [12369] = 0x0003, [12370] = 0x0100, [12372] = 0x0060, - [12373] = 0x003b, [12374] = 0x0502, [12375] = 0x0100, [12376] = 0x2fff, - [12380] = 0x2fff, [12384] = 0x2fff, [12388] = 0x2fff, [12392] = 0x2fff, - [12396] = 0x2fff, [12400] = 0x2fff, [12404] = 0x2fff, [12408] = 0x2fff, - [12412] = 0x2fff, [12416] = 0x0001, [12417] = 0x0001, [12419] = 0x0006, - [12420] = 0x0040, [12421] = 0x0001, [12422] = 0x000f, [12423] = 0x0006, - [12424] = 0x0001, [12425] = 0x0003, [12426] = 0x0106, [12428] = 0x0061, - [12429] = 0x0011, [12430] = 0x0401, [12432] = 0x0050, [12433] = 0x0003, - [12434] = 0x0102, [12436] = 0x0051, [12437] = 0x0003, [12438] = 0x0106, - [12439] = 0x000a, [12440] = 0x0061, [12441] = 0x003b, [12442] = 0x0502, - [12443] = 0x0100, [12444] = 0x2fff, [12448] = 0x2fff, [12452] = 0x2fff, - [12456] = 0x2fff, [12460] = 0x2fff, [12464] = 0x2fff, [12468] = 0x2fff, - [12472] = 0x2fff, [12476] = 0x2fff, [12480] = 0x0001, [12481] = 0x0001, - [12483] = 0x0006, [12484] = 0x0040, [12485] = 0x0001, [12486] = 0x000f, - [12487] = 0x0006, [12488] = 0x0001, [12489] = 0x0003, [12490] = 0x0106, - [12492] = 0x0061, [12493] = 0x0011, [12494] = 0x0401, [12496] = 0x0050, - [12497] = 0x0003, [12498] = 0x0102, [12500] = 0x0061, [12501] = 0x003b, - [12502] = 0x0502, [12503] = 0x0100, [12504] = 0x2fff, [12508] = 0x2fff, - [12512] = 0x2fff, [12516] = 0x2fff, [12520] = 0x2fff, [12524] = 0x2fff, - [12528] = 0x2fff, [12532] = 0x2fff, [12536] = 0x2fff, [12540] = 0x2fff, - [12544] = 0x0060, [12546] = 0x0601, [12548] = 0x0050, [12550] = 0x0100, - [12552] = 0x0001, [12554] = 0x0104, [12555] = 0x0100, [12556] = 0x2fff, - [12560] = 0x2fff, [12564] = 0x2fff, [12568] = 0x2fff, [12572] = 0x2fff, - [12576] = 0x2fff, [12580] = 0x2fff, [12584] = 0x2fff, [12588] = 0x2fff, - [12592] = 0x2fff, [12596] = 0x2fff, [12600] = 0x2fff, [12604] = 0x2fff, - [12608] = 0x0061, [12610] = 0x0601, [12612] = 0x0050, [12614] = 0x0102, - [12616] = 0x0001, [12618] = 0x0106, [12619] = 0x0100, [12620] = 0x2fff, - [12624] = 0x2fff, [12628] = 0x2fff, [12632] = 0x2fff, [12636] = 0x2fff, - [12640] = 0x2fff, [12644] = 0x2fff, [12648] = 0x2fff, [12652] = 0x2fff, - [12656] = 0x2fff, [12660] = 0x2fff, [12664] = 0x2fff, [12668] = 0x2fff, - [12672] = 0x0060, [12674] = 0x0601, [12676] = 0x0061, [12678] = 0x0601, - [12680] = 0x0050, [12682] = 0x0300, [12684] = 0x0001, [12686] = 0x0304, - [12688] = 0x0040, [12690] = 0x000f, [12692] = 0x0001, [12695] = 0x0100 +static struct reg_default wm8995_reg_defaults[] = { + { 0, 0x8995 }, + { 5, 0x0100 }, + { 16, 0x000b }, + { 17, 0x000b }, + { 24, 0x02c0 }, + { 25, 0x02c0 }, + { 26, 0x02c0 }, + { 27, 0x02c0 }, + { 28, 0x000f }, + { 32, 0x0005 }, + { 33, 0x0005 }, + { 40, 0x0003 }, + { 41, 0x0013 }, + { 48, 0x0004 }, + { 56, 0x09f8 }, + { 64, 0x1f25 }, + { 69, 0x0004 }, + { 82, 0xaaaa }, + { 84, 0x2a2a }, + { 146, 0x0060 }, + { 256, 0x0002 }, + { 257, 0x8004 }, + { 520, 0x0010 }, + { 528, 0x0083 }, + { 529, 0x0083 }, + { 548, 0x0c80 }, + { 580, 0x0c80 }, + { 768, 0x4050 }, + { 769, 0x4000 }, + { 771, 0x0040 }, + { 772, 0x0040 }, + { 773, 0x0040 }, + { 774, 0x0004 }, + { 775, 0x0100 }, + { 784, 0x4050 }, + { 785, 0x4000 }, + { 787, 0x0040 }, + { 788, 0x0040 }, + { 789, 0x0040 }, + { 1024, 0x00c0 }, + { 1025, 0x00c0 }, + { 1026, 0x00c0 }, + { 1027, 0x00c0 }, + { 1028, 0x00c0 }, + { 1029, 0x00c0 }, + { 1030, 0x00c0 }, + { 1031, 0x00c0 }, + { 1056, 0x0200 }, + { 1057, 0x0010 }, + { 1058, 0x0200 }, + { 1059, 0x0010 }, + { 1088, 0x0098 }, + { 1089, 0x0845 }, + { 1104, 0x0098 }, + { 1105, 0x0845 }, + { 1152, 0x6318 }, + { 1153, 0x6300 }, + { 1154, 0x0fca }, + { 1155, 0x0400 }, + { 1156, 0x00d8 }, + { 1157, 0x1eb5 }, + { 1158, 0xf145 }, + { 1159, 0x0b75 }, + { 1160, 0x01c5 }, + { 1161, 0x1c58 }, + { 1162, 0xf373 }, + { 1163, 0x0a54 }, + { 1164, 0x0558 }, + { 1165, 0x168e }, + { 1166, 0xf829 }, + { 1167, 0x07ad }, + { 1168, 0x1103 }, + { 1169, 0x0564 }, + { 1170, 0x0559 }, + { 1171, 0x4000 }, + { 1184, 0x6318 }, + { 1185, 0x6300 }, + { 1186, 0x0fca }, + { 1187, 0x0400 }, + { 1188, 0x00d8 }, + { 1189, 0x1eb5 }, + { 1190, 0xf145 }, + { 1191, 0x0b75 }, + { 1192, 0x01c5 }, + { 1193, 0x1c58 }, + { 1194, 0xf373 }, + { 1195, 0x0a54 }, + { 1196, 0x0558 }, + { 1197, 0x168e }, + { 1198, 0xf829 }, + { 1199, 0x07ad }, + { 1200, 0x1103 }, + { 1201, 0x0564 }, + { 1202, 0x0559 }, + { 1203, 0x4000 }, + { 1280, 0x00c0 }, + { 1281, 0x00c0 }, + { 1282, 0x00c0 }, + { 1283, 0x00c0 }, + { 1312, 0x0200 }, + { 1313, 0x0010 }, + { 1344, 0x0098 }, + { 1345, 0x0845 }, + { 1408, 0x6318 }, + { 1409, 0x6300 }, + { 1410, 0x0fca }, + { 1411, 0x0400 }, + { 1412, 0x00d8 }, + { 1413, 0x1eb5 }, + { 1414, 0xf145 }, + { 1415, 0x0b75 }, + { 1416, 0x01c5 }, + { 1417, 0x1c58 }, + { 1418, 0xf373 }, + { 1419, 0x0a54 }, + { 1420, 0x0558 }, + { 1421, 0x168e }, + { 1422, 0xf829 }, + { 1423, 0x07ad }, + { 1424, 0x1103 }, + { 1425, 0x0564 }, + { 1426, 0x0559 }, + { 1427, 0x4000 }, + { 1568, 0x0002 }, + { 1792, 0xa100 }, + { 1793, 0xa101 }, + { 1794, 0xa101 }, + { 1795, 0xa101 }, + { 1796, 0xa101 }, + { 1797, 0xa101 }, + { 1798, 0xa101 }, + { 1799, 0xa101 }, + { 1800, 0xa101 }, + { 1801, 0xa101 }, + { 1802, 0xa101 }, + { 1803, 0xa101 }, + { 1804, 0xa101 }, + { 1805, 0xa101 }, + { 1825, 0x0055 }, + { 1848, 0x3fff }, + { 1849, 0x1fff }, + { 2049, 0x0001 }, + { 2050, 0x0069 }, + { 2056, 0x0002 }, + { 2057, 0x0003 }, + { 2058, 0x0069 }, + { 12288, 0x0001 }, + { 12289, 0x0001 }, + { 12291, 0x0006 }, + { 12292, 0x0040 }, + { 12293, 0x0001 }, + { 12294, 0x000f }, + { 12295, 0x0006 }, + { 12296, 0x0001 }, + { 12297, 0x0003 }, + { 12298, 0x0104 }, + { 12300, 0x0060 }, + { 12301, 0x0011 }, + { 12302, 0x0401 }, + { 12304, 0x0050 }, + { 12305, 0x0003 }, + { 12306, 0x0100 }, + { 12308, 0x0051 }, + { 12309, 0x0003 }, + { 12310, 0x0104 }, + { 12311, 0x000a }, + { 12312, 0x0060 }, + { 12313, 0x003b }, + { 12314, 0x0502 }, + { 12315, 0x0100 }, + { 12316, 0x2fff }, + { 12320, 0x2fff }, + { 12324, 0x2fff }, + { 12328, 0x2fff }, + { 12332, 0x2fff }, + { 12336, 0x2fff }, + { 12340, 0x2fff }, + { 12344, 0x2fff }, + { 12348, 0x2fff }, + { 12352, 0x0001 }, + { 12353, 0x0001 }, + { 12355, 0x0006 }, + { 12356, 0x0040 }, + { 12357, 0x0001 }, + { 12358, 0x000f }, + { 12359, 0x0006 }, + { 12360, 0x0001 }, + { 12361, 0x0003 }, + { 12362, 0x0104 }, + { 12364, 0x0060 }, + { 12365, 0x0011 }, + { 12366, 0x0401 }, + { 12368, 0x0050 }, + { 12369, 0x0003 }, + { 12370, 0x0100 }, + { 12372, 0x0060 }, + { 12373, 0x003b }, + { 12374, 0x0502 }, + { 12375, 0x0100 }, + { 12376, 0x2fff }, + { 12380, 0x2fff }, + { 12384, 0x2fff }, + { 12388, 0x2fff }, + { 12392, 0x2fff }, + { 12396, 0x2fff }, + { 12400, 0x2fff }, + { 12404, 0x2fff }, + { 12408, 0x2fff }, + { 12412, 0x2fff }, + { 12416, 0x0001 }, + { 12417, 0x0001 }, + { 12419, 0x0006 }, + { 12420, 0x0040 }, + { 12421, 0x0001 }, + { 12422, 0x000f }, + { 12423, 0x0006 }, + { 12424, 0x0001 }, + { 12425, 0x0003 }, + { 12426, 0x0106 }, + { 12428, 0x0061 }, + { 12429, 0x0011 }, + { 12430, 0x0401 }, + { 12432, 0x0050 }, + { 12433, 0x0003 }, + { 12434, 0x0102 }, + { 12436, 0x0051 }, + { 12437, 0x0003 }, + { 12438, 0x0106 }, + { 12439, 0x000a }, + { 12440, 0x0061 }, + { 12441, 0x003b }, + { 12442, 0x0502 }, + { 12443, 0x0100 }, + { 12444, 0x2fff }, + { 12448, 0x2fff }, + { 12452, 0x2fff }, + { 12456, 0x2fff }, + { 12460, 0x2fff }, + { 12464, 0x2fff }, + { 12468, 0x2fff }, + { 12472, 0x2fff }, + { 12476, 0x2fff }, + { 12480, 0x0001 }, + { 12481, 0x0001 }, + { 12483, 0x0006 }, + { 12484, 0x0040 }, + { 12485, 0x0001 }, + { 12486, 0x000f }, + { 12487, 0x0006 }, + { 12488, 0x0001 }, + { 12489, 0x0003 }, + { 12490, 0x0106 }, + { 12492, 0x0061 }, + { 12493, 0x0011 }, + { 12494, 0x0401 }, + { 12496, 0x0050 }, + { 12497, 0x0003 }, + { 12498, 0x0102 }, + { 12500, 0x0061 }, + { 12501, 0x003b }, + { 12502, 0x0502 }, + { 12503, 0x0100 }, + { 12504, 0x2fff }, + { 12508, 0x2fff }, + { 12512, 0x2fff }, + { 12516, 0x2fff }, + { 12520, 0x2fff }, + { 12524, 0x2fff }, + { 12528, 0x2fff }, + { 12532, 0x2fff }, + { 12536, 0x2fff }, + { 12540, 0x2fff }, + { 12544, 0x0060 }, + { 12546, 0x0601 }, + { 12548, 0x0050 }, + { 12550, 0x0100 }, + { 12552, 0x0001 }, + { 12554, 0x0104 }, + { 12555, 0x0100 }, + { 12556, 0x2fff }, + { 12560, 0x2fff }, + { 12564, 0x2fff }, + { 12568, 0x2fff }, + { 12572, 0x2fff }, + { 12576, 0x2fff }, + { 12580, 0x2fff }, + { 12584, 0x2fff }, + { 12588, 0x2fff }, + { 12592, 0x2fff }, + { 12596, 0x2fff }, + { 12600, 0x2fff }, + { 12604, 0x2fff }, + { 12608, 0x0061 }, + { 12610, 0x0601 }, + { 12612, 0x0050 }, + { 12614, 0x0102 }, + { 12616, 0x0001 }, + { 12618, 0x0106 }, + { 12619, 0x0100 }, + { 12620, 0x2fff }, + { 12624, 0x2fff }, + { 12628, 0x2fff }, + { 12632, 0x2fff }, + { 12636, 0x2fff }, + { 12640, 0x2fff }, + { 12644, 0x2fff }, + { 12648, 0x2fff }, + { 12652, 0x2fff }, + { 12656, 0x2fff }, + { 12660, 0x2fff }, + { 12664, 0x2fff }, + { 12668, 0x2fff }, + { 12672, 0x0060 }, + { 12674, 0x0601 }, + { 12676, 0x0061 }, + { 12678, 0x0601 }, + { 12680, 0x0050 }, + { 12682, 0x0300 }, + { 12684, 0x0001 }, + { 12686, 0x0304 }, + { 12688, 0x0040 }, + { 12690, 0x000f }, + { 12692, 0x0001 }, + { 12695, 0x0100 }, }; struct fll_config { @@ -134,7 +378,7 @@ struct fll_config { }; struct wm8995_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk[2]; int mclk[2]; int aifclk[2]; @@ -156,7 +400,7 @@ static int wm8995_regulator_event_##n(struct notifier_block *nb, \ struct wm8995_priv *wm8995 = container_of(nb, struct wm8995_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - wm8995->codec->cache_sync = 1; \ + regcache_mark_dirty(wm8995->regmap); \ } \ return 0; \ } @@ -949,31 +1193,244 @@ static const struct snd_soc_dapm_route wm8995_intercon[] = { { "SPK2R", NULL, "SPK2R Driver" } }; -static int wm8995_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8995_readable(struct device *dev, unsigned int reg) { - /* out of bounds registers are generally considered - * volatile to support register banks that are partially - * owned by something else for e.g. a DSP - */ - if (reg > WM8995_MAX_CACHED_REGISTER) - return 1; - switch (reg) { case WM8995_SOFTWARE_RESET: + case WM8995_POWER_MANAGEMENT_1: + case WM8995_POWER_MANAGEMENT_2: + case WM8995_POWER_MANAGEMENT_3: + case WM8995_POWER_MANAGEMENT_4: + case WM8995_POWER_MANAGEMENT_5: + case WM8995_LEFT_LINE_INPUT_1_VOLUME: + case WM8995_RIGHT_LINE_INPUT_1_VOLUME: + case WM8995_LEFT_LINE_INPUT_CONTROL: + case WM8995_DAC1_LEFT_VOLUME: + case WM8995_DAC1_RIGHT_VOLUME: + case WM8995_DAC2_LEFT_VOLUME: + case WM8995_DAC2_RIGHT_VOLUME: + case WM8995_OUTPUT_VOLUME_ZC_1: + case WM8995_MICBIAS_1: + case WM8995_MICBIAS_2: + case WM8995_LDO_1: + case WM8995_LDO_2: + case WM8995_ACCESSORY_DETECT_MODE1: + case WM8995_ACCESSORY_DETECT_MODE2: + case WM8995_HEADPHONE_DETECT1: + case WM8995_HEADPHONE_DETECT2: + case WM8995_MIC_DETECT_1: + case WM8995_MIC_DETECT_2: + case WM8995_CHARGE_PUMP_1: + case WM8995_CLASS_W_1: + case WM8995_DC_SERVO_1: + case WM8995_DC_SERVO_2: + case WM8995_DC_SERVO_3: + case WM8995_DC_SERVO_5: + case WM8995_DC_SERVO_6: + case WM8995_DC_SERVO_7: case WM8995_DC_SERVO_READBACK_0: + case WM8995_ANALOGUE_HP_1: + case WM8995_ANALOGUE_HP_2: + case WM8995_CHIP_REVISION: + case WM8995_CONTROL_INTERFACE_1: + case WM8995_CONTROL_INTERFACE_2: + case WM8995_WRITE_SEQUENCER_CTRL_1: + case WM8995_WRITE_SEQUENCER_CTRL_2: + case WM8995_AIF1_CLOCKING_1: + case WM8995_AIF1_CLOCKING_2: + case WM8995_AIF2_CLOCKING_1: + case WM8995_AIF2_CLOCKING_2: + case WM8995_CLOCKING_1: + case WM8995_CLOCKING_2: + case WM8995_AIF1_RATE: + case WM8995_AIF2_RATE: + case WM8995_RATE_STATUS: + case WM8995_FLL1_CONTROL_1: + case WM8995_FLL1_CONTROL_2: + case WM8995_FLL1_CONTROL_3: + case WM8995_FLL1_CONTROL_4: + case WM8995_FLL1_CONTROL_5: + case WM8995_FLL2_CONTROL_1: + case WM8995_FLL2_CONTROL_2: + case WM8995_FLL2_CONTROL_3: + case WM8995_FLL2_CONTROL_4: + case WM8995_FLL2_CONTROL_5: + case WM8995_AIF1_CONTROL_1: + case WM8995_AIF1_CONTROL_2: + case WM8995_AIF1_MASTER_SLAVE: + case WM8995_AIF1_BCLK: + case WM8995_AIF1ADC_LRCLK: + case WM8995_AIF1DAC_LRCLK: + case WM8995_AIF1DAC_DATA: + case WM8995_AIF1ADC_DATA: + case WM8995_AIF2_CONTROL_1: + case WM8995_AIF2_CONTROL_2: + case WM8995_AIF2_MASTER_SLAVE: + case WM8995_AIF2_BCLK: + case WM8995_AIF2ADC_LRCLK: + case WM8995_AIF2DAC_LRCLK: + case WM8995_AIF2DAC_DATA: + case WM8995_AIF2ADC_DATA: + case WM8995_AIF1_ADC1_LEFT_VOLUME: + case WM8995_AIF1_ADC1_RIGHT_VOLUME: + case WM8995_AIF1_DAC1_LEFT_VOLUME: + case WM8995_AIF1_DAC1_RIGHT_VOLUME: + case WM8995_AIF1_ADC2_LEFT_VOLUME: + case WM8995_AIF1_ADC2_RIGHT_VOLUME: + case WM8995_AIF1_DAC2_LEFT_VOLUME: + case WM8995_AIF1_DAC2_RIGHT_VOLUME: + case WM8995_AIF1_ADC1_FILTERS: + case WM8995_AIF1_ADC2_FILTERS: + case WM8995_AIF1_DAC1_FILTERS_1: + case WM8995_AIF1_DAC1_FILTERS_2: + case WM8995_AIF1_DAC2_FILTERS_1: + case WM8995_AIF1_DAC2_FILTERS_2: + case WM8995_AIF1_DRC1_1: + case WM8995_AIF1_DRC1_2: + case WM8995_AIF1_DRC1_3: + case WM8995_AIF1_DRC1_4: + case WM8995_AIF1_DRC1_5: + case WM8995_AIF1_DRC2_1: + case WM8995_AIF1_DRC2_2: + case WM8995_AIF1_DRC2_3: + case WM8995_AIF1_DRC2_4: + case WM8995_AIF1_DRC2_5: + case WM8995_AIF1_DAC1_EQ_GAINS_1: + case WM8995_AIF1_DAC1_EQ_GAINS_2: + case WM8995_AIF1_DAC1_EQ_BAND_1_A: + case WM8995_AIF1_DAC1_EQ_BAND_1_B: + case WM8995_AIF1_DAC1_EQ_BAND_1_PG: + case WM8995_AIF1_DAC1_EQ_BAND_2_A: + case WM8995_AIF1_DAC1_EQ_BAND_2_B: + case WM8995_AIF1_DAC1_EQ_BAND_2_C: + case WM8995_AIF1_DAC1_EQ_BAND_2_PG: + case WM8995_AIF1_DAC1_EQ_BAND_3_A: + case WM8995_AIF1_DAC1_EQ_BAND_3_B: + case WM8995_AIF1_DAC1_EQ_BAND_3_C: + case WM8995_AIF1_DAC1_EQ_BAND_3_PG: + case WM8995_AIF1_DAC1_EQ_BAND_4_A: + case WM8995_AIF1_DAC1_EQ_BAND_4_B: + case WM8995_AIF1_DAC1_EQ_BAND_4_C: + case WM8995_AIF1_DAC1_EQ_BAND_4_PG: + case WM8995_AIF1_DAC1_EQ_BAND_5_A: + case WM8995_AIF1_DAC1_EQ_BAND_5_B: + case WM8995_AIF1_DAC1_EQ_BAND_5_PG: + case WM8995_AIF1_DAC2_EQ_GAINS_1: + case WM8995_AIF1_DAC2_EQ_GAINS_2: + case WM8995_AIF1_DAC2_EQ_BAND_1_A: + case WM8995_AIF1_DAC2_EQ_BAND_1_B: + case WM8995_AIF1_DAC2_EQ_BAND_1_PG: + case WM8995_AIF1_DAC2_EQ_BAND_2_A: + case WM8995_AIF1_DAC2_EQ_BAND_2_B: + case WM8995_AIF1_DAC2_EQ_BAND_2_C: + case WM8995_AIF1_DAC2_EQ_BAND_2_PG: + case WM8995_AIF1_DAC2_EQ_BAND_3_A: + case WM8995_AIF1_DAC2_EQ_BAND_3_B: + case WM8995_AIF1_DAC2_EQ_BAND_3_C: + case WM8995_AIF1_DAC2_EQ_BAND_3_PG: + case WM8995_AIF1_DAC2_EQ_BAND_4_A: + case WM8995_AIF1_DAC2_EQ_BAND_4_B: + case WM8995_AIF1_DAC2_EQ_BAND_4_C: + case WM8995_AIF1_DAC2_EQ_BAND_4_PG: + case WM8995_AIF1_DAC2_EQ_BAND_5_A: + case WM8995_AIF1_DAC2_EQ_BAND_5_B: + case WM8995_AIF1_DAC2_EQ_BAND_5_PG: + case WM8995_AIF2_ADC_LEFT_VOLUME: + case WM8995_AIF2_ADC_RIGHT_VOLUME: + case WM8995_AIF2_DAC_LEFT_VOLUME: + case WM8995_AIF2_DAC_RIGHT_VOLUME: + case WM8995_AIF2_ADC_FILTERS: + case WM8995_AIF2_DAC_FILTERS_1: + case WM8995_AIF2_DAC_FILTERS_2: + case WM8995_AIF2_DRC_1: + case WM8995_AIF2_DRC_2: + case WM8995_AIF2_DRC_3: + case WM8995_AIF2_DRC_4: + case WM8995_AIF2_DRC_5: + case WM8995_AIF2_EQ_GAINS_1: + case WM8995_AIF2_EQ_GAINS_2: + case WM8995_AIF2_EQ_BAND_1_A: + case WM8995_AIF2_EQ_BAND_1_B: + case WM8995_AIF2_EQ_BAND_1_PG: + case WM8995_AIF2_EQ_BAND_2_A: + case WM8995_AIF2_EQ_BAND_2_B: + case WM8995_AIF2_EQ_BAND_2_C: + case WM8995_AIF2_EQ_BAND_2_PG: + case WM8995_AIF2_EQ_BAND_3_A: + case WM8995_AIF2_EQ_BAND_3_B: + case WM8995_AIF2_EQ_BAND_3_C: + case WM8995_AIF2_EQ_BAND_3_PG: + case WM8995_AIF2_EQ_BAND_4_A: + case WM8995_AIF2_EQ_BAND_4_B: + case WM8995_AIF2_EQ_BAND_4_C: + case WM8995_AIF2_EQ_BAND_4_PG: + case WM8995_AIF2_EQ_BAND_5_A: + case WM8995_AIF2_EQ_BAND_5_B: + case WM8995_AIF2_EQ_BAND_5_PG: + case WM8995_DAC1_MIXER_VOLUMES: + case WM8995_DAC1_LEFT_MIXER_ROUTING: + case WM8995_DAC1_RIGHT_MIXER_ROUTING: + case WM8995_DAC2_MIXER_VOLUMES: + case WM8995_DAC2_LEFT_MIXER_ROUTING: + case WM8995_DAC2_RIGHT_MIXER_ROUTING: + case WM8995_AIF1_ADC1_LEFT_MIXER_ROUTING: + case WM8995_AIF1_ADC1_RIGHT_MIXER_ROUTING: + case WM8995_AIF1_ADC2_LEFT_MIXER_ROUTING: + case WM8995_AIF1_ADC2_RIGHT_MIXER_ROUTING: + case WM8995_DAC_SOFTMUTE: + case WM8995_OVERSAMPLING: + case WM8995_SIDETONE: + case WM8995_GPIO_1: + case WM8995_GPIO_2: + case WM8995_GPIO_3: + case WM8995_GPIO_4: + case WM8995_GPIO_5: + case WM8995_GPIO_6: + case WM8995_GPIO_7: + case WM8995_GPIO_8: + case WM8995_GPIO_9: + case WM8995_GPIO_10: + case WM8995_GPIO_11: + case WM8995_GPIO_12: + case WM8995_GPIO_13: + case WM8995_GPIO_14: + case WM8995_PULL_CONTROL_1: + case WM8995_PULL_CONTROL_2: case WM8995_INTERRUPT_STATUS_1: case WM8995_INTERRUPT_STATUS_2: + case WM8995_INTERRUPT_RAW_STATUS_2: case WM8995_INTERRUPT_STATUS_1_MASK: case WM8995_INTERRUPT_STATUS_2_MASK: case WM8995_INTERRUPT_CONTROL: + case WM8995_LEFT_PDM_SPEAKER_1: + case WM8995_RIGHT_PDM_SPEAKER_1: + case WM8995_PDM_SPEAKER_1_MUTE_SEQUENCE: + case WM8995_LEFT_PDM_SPEAKER_2: + case WM8995_RIGHT_PDM_SPEAKER_2: + case WM8995_PDM_SPEAKER_2_MUTE_SEQUENCE: + return true; + default: + return false; + } +} + +static bool wm8995_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8995_SOFTWARE_RESET: + case WM8995_DC_SERVO_READBACK_0: + case WM8995_INTERRUPT_STATUS_1: + case WM8995_INTERRUPT_STATUS_2: + case WM8995_INTERRUPT_CONTROL: case WM8995_ACCESSORY_DETECT_MODE1: case WM8995_ACCESSORY_DETECT_MODE2: case WM8995_HEADPHONE_DETECT1: case WM8995_HEADPHONE_DETECT2: - return 1; + case WM8995_RATE_STATUS: + return true; + default: + return false; } - - return 0; } static int wm8995_aif_mute(struct snd_soc_dai *dai, int mute) @@ -1528,7 +1985,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, if (ret) return ret; - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(wm8995->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -1594,7 +2051,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, wm8995->control_type); + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; @@ -1783,11 +2240,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .suspend = wm8995_suspend, .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8995_reg_defs), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8995_reg_defs, - .volatile_register = wm8995_volatile, - .compress_type = SND_SOC_RBTREE_COMPRESSION +}; + +static struct regmap_config wm8995_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = WM8995_MAX_REGISTER, + .reg_defaults = wm8995_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8995_reg_defaults), + .volatile_reg = wm8995_volatile, + .readable_reg = wm8995_readable, + .cache_type = REGCACHE_RBTREE, }; #if defined(CONFIG_SPI_MASTER) @@ -1800,21 +2264,37 @@ static int __devinit wm8995_spi_probe(struct spi_device *spi) if (!wm8995) return -ENOMEM; - wm8995->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8995); + wm8995->regmap = regmap_init_spi(spi, &wm8995_regmap); + if (IS_ERR(wm8995->regmap)) { + ret = PTR_ERR(wm8995->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + goto err_alloc; + } + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8995, wm8995_dai, ARRAY_SIZE(wm8995_dai)); if (ret < 0) - kfree(wm8995); + goto err_regmap; + + return ret; + +err_regmap: + regmap_exit(wm8995->regmap); +err_alloc: + kfree(wm8995); + return ret; } static int __devexit wm8995_spi_remove(struct spi_device *spi) { + struct wm8995_priv *wm8995 = spi_get_drvdata(spi); snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(wm8995->regmap); + kfree(wm8995); return 0; } @@ -1839,21 +2319,40 @@ static __devinit int wm8995_i2c_probe(struct i2c_client *i2c, if (!wm8995) return -ENOMEM; - wm8995->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, wm8995); + wm8995->regmap = regmap_init_i2c(i2c, &wm8995_regmap); + if (IS_ERR(wm8995->regmap)) { + ret = PTR_ERR(wm8995->regmap); + dev_err(&i2c->dev, "Failed to register regmap: %d\n", ret); + goto err_alloc; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8995, wm8995_dai, ARRAY_SIZE(wm8995_dai)); - if (ret < 0) - kfree(wm8995); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } + + return ret; + +err_regmap: + regmap_exit(wm8995->regmap); +err_alloc: + kfree(wm8995); + return ret; } static __devexit int wm8995_i2c_remove(struct i2c_client *client) { + struct wm8995_priv *wm8995 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(wm8995->regmap); + kfree(wm8995); return 0; } -- 1.7.7.2

1 0

[alsa-devel] [PATCH] ASoC: sta32x: add workaround for ESD reset issue
by Johannes Stezenbach 10 Nov '11

10 Nov '11

sta32x resets and loses all configuration during ESD test. Work around by preserving coefficient RAM in a shadow, poll once a second on the CONFA register and restore all coeffcients and registers when CONFA changes unexpectedly. Cc: Sven Neumann <s.neumann(a)raumfeld.com> Cc: Daniel Mack <daniel(a)zonque.org> Signed-off-by: Johannes Stezenbach <js(a)sig21.net> --- ESD test means they are shooting the poor thing with a taser and then sneer at the weakling when it keels over in shock. You wouldn't believe what atrocities some "engineers" are capable of, all in the name of "quality assurance". The patch applies cleanly against both for-3.2 and for-3.3 of git://opensource.wolfsonmicro.com/linux-2.6-asoc.git. sound/soc/codecs/sta32x.c | 75 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 1 + 2 files changed, 76 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index bb82408..3caa7c3 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -27,6 +27,7 @@ #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> +#include <linux/workqueue.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -76,6 +77,10 @@ struct sta32x_priv { unsigned int mclk; unsigned int format; + + u32 coef_shadow[STA32X_COEF_COUNT]; + struct delayed_work watchdog_work; + int shutdown; }; static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); @@ -227,6 +232,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); int numcoef = kcontrol->private_value >> 16; int index = kcontrol->private_value & 0xffff; unsigned int cfud; @@ -239,6 +245,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, snd_soc_write(codec, STA32X_CFUD, cfud); snd_soc_write(codec, STA32X_CFADDR2, index); + for (i = 0; i < numcoef && (index + (i + 1) * 3 < STA32X_COEF_COUNT); i++) + sta32x->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i ] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); for (i = 0; i < 3 * numcoef; i++) snd_soc_write(codec, STA32X_B1CF1 + i, ucontrol->value.bytes.data[i]); @@ -252,6 +263,55 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, return 0; } +/* work around ESD issue where sta32x resets and loses all configuration */ +static void sta32x_watchdog(struct work_struct *work) +{ + struct sta32x_priv *sta32x = container_of(work, struct sta32x_priv, + watchdog_work.work); + struct snd_soc_codec *codec = sta32x->codec; + unsigned int value; + unsigned int cfud; + unsigned int mute; + unsigned int confa; + int i; + + /* check if sta32x has reset itself */ + snd_soc_cache_read(codec, STA32X_CONFA, &confa); + if (confa == codec->hw_read(codec, STA32X_CONFA)) + goto ok; + + /* mute during restore */ + snd_soc_cache_read(codec, STA32X_MMUTE, &mute); + snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE); + + for (i = 0; i < STA32X_REGISTER_COUNT; i++) { + if (snd_soc_codec_volatile_register(codec, i)) + continue; + snd_soc_cache_read(codec, i, &value); + snd_soc_write(codec, i, value); + } + + /* preserve reserved bits in STA32X_CFUD */ + cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0; + + for (i = 0; i < STA32X_COEF_COUNT; i++) { + snd_soc_write(codec, STA32X_CFADDR2, i); + snd_soc_write(codec, STA32X_B1CF1, (sta32x->coef_shadow[i] >> 16) & 0xff); + snd_soc_write(codec, STA32X_B1CF2, (sta32x->coef_shadow[i] >> 8) & 0xff); + snd_soc_write(codec, STA32X_B1CF3, (sta32x->coef_shadow[i] ) & 0xff); + /* chip documentation does not say if the bits are self clearing, + * so do it explicitly */ + snd_soc_write(codec, STA32X_CFUD, cfud); + snd_soc_write(codec, STA32X_CFUD, cfud | 0x01); + } + snd_soc_write(codec, STA32X_MMUTE, mute); + +ok: + if (!sta32x->shutdown) + schedule_delayed_work(&sta32x->watchdog_work, + round_jiffies_relative(HZ)); +} + #define SINGLE_COEF(xname, index) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = sta32x_coefficient_info, \ @@ -790,10 +850,23 @@ static int sta32x_probe(struct snd_soc_codec *codec) STA32X_CxCFG_OM_MASK, 2 << STA32X_CxCFG_OM_SHIFT); + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta32x->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i ++) + sta32x->coef_shadow[i] = 0x7fffff; + sta32x->coef_shadow[55] = 0x5a9df7; + sta32x->coef_shadow[56] = 0x7fffff; + sta32x->coef_shadow[59] = 0x7fffff; + sta32x->coef_shadow[60] = 0x400000; + sta32x->coef_shadow[61] = 0x400000; + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + INIT_DELAYED_WORK(&sta32x->watchdog_work, sta32x_watchdog); + schedule_delayed_work(&sta32x->watchdog_work, round_jiffies_relative(HZ)); return 0; err_get: @@ -806,6 +879,8 @@ static int sta32x_remove(struct snd_soc_codec *codec) { struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + sta32x->shutdown = 1; + cancel_delayed_work_sync(&sta32x->watchdog_work); sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h index b97ee5a..d8e32a6 100644 --- a/sound/soc/codecs/sta32x.h +++ b/sound/soc/codecs/sta32x.h @@ -19,6 +19,7 @@ /* STA326 register addresses */ #define STA32X_REGISTER_COUNT 0x2d +#define STA32X_COEF_COUNT 62 #define STA32X_CONFA 0x00 #define STA32X_CONFB 0x01 -- 1.7.7.2

3 6

[alsa-devel] iec958 device
by Vipin Kumar 10 Nov '11

10 Nov '11

Hello, Sorry if it seems like a basic question but I am not able to comprehend it completely I am writing an SPDIF controller's driver which is almost complete I understand that a user should be able to see a iec958 device with "alsa -L" I am not able to comprehend how to do it. Is it just a user configuration or something needs to be exported from the driver as well Thanks in advance Regards Vipin

2 1

[alsa-devel] [PATCH] ASoC: Move WM5100 platform data based setup into I2C probe
by Mark Brown 10 Nov '11

10 Nov '11

Get things configured as early as possible, especially useful for the GPIOs which might be useful anyway. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 36 ++++++++++++++++++------------------ 1 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index f72de26..03cb0b0 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2580,24 +2580,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, WM5100_OUT_VU); - for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { - snd_soc_update_bits(codec, WM5100_IN1L_CONTROL, - WM5100_IN1_MODE_MASK | - WM5100_IN1_DMIC_SUP_MASK, - (wm5100->pdata.in_mode[i] << - WM5100_IN1_MODE_SHIFT) | - (wm5100->pdata.dmic_sup[i] << - WM5100_IN1_DMIC_SUP_SHIFT)); - } - - for (i = 0; i < ARRAY_SIZE(wm5100->pdata.gpio_defaults); i++) { - if (!wm5100->pdata.gpio_defaults[i]) - continue; - - snd_soc_write(codec, WM5100_GPIO_CTRL_1 + i, - wm5100->pdata.gpio_defaults[i]); - } - /* Don't debounce interrupts to support use of SYSCLK only */ snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_1, 0); snd_soc_write(codec, WM5100_IRQ_DEBOUNCE_2, 0); @@ -2862,6 +2844,24 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, wm5100_init_gpio(i2c); + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.gpio_defaults); i++) { + if (!wm5100->pdata.gpio_defaults[i]) + continue; + + regmap_write(wm5100->regmap, WM5100_GPIO_CTRL_1 + i, + wm5100->pdata.gpio_defaults[i]); + } + + for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { + regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL, + WM5100_IN1_MODE_MASK | + WM5100_IN1_DMIC_SUP_MASK, + (wm5100->pdata.in_mode[i] << + WM5100_IN1_MODE_SHIFT) | + (wm5100->pdata.dmic_sup[i] << + WM5100_IN1_DMIC_SUP_SHIFT)); + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); -- 1.7.7.2

1 0

[alsa-devel] Success with TASCAM US144 MKII, linux-3.0
by Matteo Frigo 10 Nov '11

10 Nov '11

Dear ALSA developers, The purpose of this email is to report that I was able to play music with a TASCAM US144 MKII using linux-3.0, specifically linux-3.0.0-2-amd64 from Debian wheezy. Various reports on this mailing list and elsewhere seem to suggest that the US144 works with ALSA but that the US144-MKII does not work. I want to confirm that the US144-MKII does in fact work with snd_usb_us122l, at least in playback mode. Here is what I did. *Before* inserting the USB cable, execute "rmmod ehci-hcd". It seems like the unit uses a different protocol in USB2 mode, and I guess it must be plugged into a USB1 interface. The unit does not work if booted in USB2 mode and then reset to USB1. Add these lines to /etc/asound.conf: pcm.!usb_stream { type usb_stream card "1" } ctl.!usb_stream { type hw card 1 } Then aplay --buffer-size=1000 -MDplug:usb_stream foo.wav works for me, where foo.wav is a {16,24}-bit stereo {44.1,48}-KHz PCM file. I am not sure whether the --buffer-size=1000 is necessary---I am using a slow Atom netbook and the sound is choppy with the default buffer size. Increasing the buffer size seems to fix the problem. I don't know if the firmware revision matters, but my US144-MKII unit has the lastest firmware 2.03, which I installed via a Windows machine. Regards, Matteo Frigo

1 0

[alsa-devel] [PATCH] ASoC don't clobber platform DMA driver ops
by Alan Tull 10 Nov '11

10 Nov '11

Previously, only one static struct for ops existed for all platform DMA drivers to share. Half of the ops are shared functions which don't have stubs in the ALSA core. The other half are initialized to point directly to ops in the platform driver. This creates problems where each time soc_new_pcm is called, the new platform driver's ops would overwrite a subset of the ops. This patch creates and populates a ops struct for each call to soc_pcm_new. Also creates a common soc_pcm_ioctl function that calls all the driver components for ioctl calls. Signed-off-by: Alan Tull <alan.tull(a)freescale.com> --- include/sound/soc-dai.h | 2 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 80 +++++++++++++++++++++++++++++++++++----------- 4 files changed, 65 insertions(+), 19 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5ad5f3a..b3f880e 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -172,6 +172,8 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); + int (*ioctl)(struct snd_pcm_substream *, unsigned int, void *, + struct snd_soc_dai *); /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..1574325 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -517,6 +517,7 @@ struct snd_soc_ops { int (*hw_free)(struct snd_pcm_substream *); int (*prepare)(struct snd_pcm_substream *); int (*trigger)(struct snd_pcm_substream *, int); + int (*ioctl)(struct snd_pcm_substream *, unsigned int, void *); }; /* SoC cache ops */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b085d8e..1b48fc6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1547,6 +1547,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_free(&card->dapm); + kfree(card->rtd->ops); kfree(card->rtd); snd_card_free(card->snd_card); return 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2879c88..2bd7504 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -567,16 +567,39 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; +static int soc_pcm_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + if (rtd->dai_link->ops && rtd->dai_link->ops->ioctl) { + ret = rtd->dai_link->ops->ioctl(substream, cmd, arg); + if (ret < 0) + return ret; + } + + if (codec_dai->driver->ops->ioctl) { + ret = codec_dai->driver->ops->ioctl(substream, cmd, arg, codec_dai); + if (ret < 0) + return ret; + } + + if (platform->driver->ops->ioctl) { + ret = platform->driver->ops->ioctl(substream, cmd, arg); + if (ret < 0) + return ret; + } + + if (cpu_dai->driver->ops->ioctl) { + ret = cpu_dai->driver->ops->ioctl(substream, cmd, arg, cpu_dai); + if (ret < 0) + return ret; + } + return 0; +} /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) @@ -586,9 +609,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; + struct snd_pcm_ops *soc_pcm_ops; char new_name[64]; int ret = 0, playback = 0, capture = 0; + /* ASoC PCM operations */ + soc_pcm_ops = kzalloc(sizeof(struct snd_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + printk(KERN_ERR "asoc: can't create pcm ops for codec %s\n", codec->name); + return -ENOMEM; + } + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -603,34 +634,45 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) num, playback, capture, &pcm); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + kfree(soc_pcm_ops); return ret; } /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_codec_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->pointer = soc_pcm_pointer; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->ioctl = soc_pcm_ioctl; + rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } + rtd->ops = soc_pcm_ops; + if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { pr_err("asoc: platform pcm constructor failed\n"); + kfree(rtd->ops); return ret; } } -- 1.6.0.4

4 7

[alsa-devel] Playback and capture synchronization
by anish kumar 10 Nov '11

10 Nov '11

> > Patrick Lai wrote: > >> I have a use case requiring playback and capture starting at the same > >> time. Latency between starting of playback and capture streams need to > >> be less than few PCM samples. Base on my understanding on ALSA which > >> does not have the concept of grouping multiple streams, Is this use case for loopback?Right now i am doing loopback with two terminals with one terminal running aplay and other arecord and before running aplay and arecord we use couple of mixer controls to set the paths but we want to do all this from user space i.e. ALSA UCM(use case manager) & Android Audio Flinger.We are not using snd-aloop for loopback. After reading this thread looks like I need to implement support in driver to trigger playback and capture at the same time using the api's mentioned in this thread. Is my understanding correct?Or am I completely off track here? Kernel used is:3.0 > > snd_pcm_link() > Awesome! I will look into this API > > > If the driver doesn't actually implement starting of linked streams > > (which would be pointless if the hardware doesn't support it), ALSA > > calls the second trigger callback after the first one. > > > > It might be possible to implement simultaneous starting of two streams. > > What hardware (driver) are you using? > It's Qualcomm hardware. As far as I know, HW has this synchronization > support but I would have to build driver to make use of this functionality > To see examples of drivers that support HW linking, search for > snd_pcm_group_for_each_entry

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[alsa-devel] [PATCH] ASoC: Remove LZO cache type
by Mark Brown 10 Nov '11

10 Nov '11

There are no current users and new drivers ought to be using the regmap API and its cache implementation directly so just delete the ASoC copy. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 - sound/soc/Kconfig | 15 -- sound/soc/soc-cache.c | 384 ------------------------------------------------- 3 files changed, 0 insertions(+), 400 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index c39ac19..c12cee6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -273,7 +273,6 @@ enum snd_soc_control_type { enum snd_soc_compress_type { SND_SOC_FLAT_COMPRESSION = 1, - SND_SOC_LZO_COMPRESSION, SND_SOC_RBTREE_COMPRESSION }; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 1381db8..35e662d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -22,21 +22,6 @@ menuconfig SND_SOC if SND_SOC -config SND_SOC_CACHE_LZO - bool "Support LZO compression for register caches" - select LZO_COMPRESS - select LZO_DECOMPRESS - ---help--- - Select this to enable LZO compression for register caches. - This will allow machine or CODEC drivers to compress register - caches in memory, reducing the memory consumption at the - expense of performance. If this is not present and is used - the system will fall back to uncompressed caches. - - Usually it is safe to disable this option, where cache - compression in used the rbtree option will typically perform - better. - config SND_SOC_AC97_BUS bool diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9077aa4..18bb6b3 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -14,7 +14,6 @@ #include <linux/i2c.h> #include <linux/spi/spi.h> #include <sound/soc.h> -#include <linux/lzo.h> #include <linux/bitmap.h> #include <linux/rbtree.h> #include <linux/export.h> @@ -439,378 +438,6 @@ err: return ret; } -#ifdef CONFIG_SND_SOC_CACHE_LZO -struct snd_soc_lzo_ctx { - void *wmem; - void *dst; - const void *src; - size_t src_len; - size_t dst_len; - size_t decompressed_size; - unsigned long *sync_bmp; - int sync_bmp_nbits; -}; - -#define LZO_BLOCK_NUM 8 -static int snd_soc_lzo_block_count(void) -{ - return LZO_BLOCK_NUM; -} - -static int snd_soc_lzo_prepare(struct snd_soc_lzo_ctx *lzo_ctx) -{ - lzo_ctx->wmem = kmalloc(LZO1X_MEM_COMPRESS, GFP_KERNEL); - if (!lzo_ctx->wmem) - return -ENOMEM; - return 0; -} - -static int snd_soc_lzo_compress(struct snd_soc_lzo_ctx *lzo_ctx) -{ - size_t compress_size; - int ret; - - ret = lzo1x_1_compress(lzo_ctx->src, lzo_ctx->src_len, - lzo_ctx->dst, &compress_size, lzo_ctx->wmem); - if (ret != LZO_E_OK || compress_size > lzo_ctx->dst_len) - return -EINVAL; - lzo_ctx->dst_len = compress_size; - return 0; -} - -static int snd_soc_lzo_decompress(struct snd_soc_lzo_ctx *lzo_ctx) -{ - size_t dst_len; - int ret; - - dst_len = lzo_ctx->dst_len; - ret = lzo1x_decompress_safe(lzo_ctx->src, lzo_ctx->src_len, - lzo_ctx->dst, &dst_len); - if (ret != LZO_E_OK || dst_len != lzo_ctx->dst_len) - return -EINVAL; - return 0; -} - -static int snd_soc_lzo_compress_cache_block(struct snd_soc_codec *codec, - struct snd_soc_lzo_ctx *lzo_ctx) -{ - int ret; - - lzo_ctx->dst_len = lzo1x_worst_compress(PAGE_SIZE); - lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); - if (!lzo_ctx->dst) { - lzo_ctx->dst_len = 0; - return -ENOMEM; - } - - ret = snd_soc_lzo_compress(lzo_ctx); - if (ret < 0) - return ret; - return 0; -} - -static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec, - struct snd_soc_lzo_ctx *lzo_ctx) -{ - int ret; - - lzo_ctx->dst_len = lzo_ctx->decompressed_size; - lzo_ctx->dst = kmalloc(lzo_ctx->dst_len, GFP_KERNEL); - if (!lzo_ctx->dst) { - lzo_ctx->dst_len = 0; - return -ENOMEM; - } - - ret = snd_soc_lzo_decompress(lzo_ctx); - if (ret < 0) - return ret; - return 0; -} - -static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - return (reg * codec_drv->reg_word_size) / - DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); -} - -static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - - codec_drv = codec->driver; - return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) / - codec_drv->reg_word_size); -} - -static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) -{ - return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); -} - -static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - unsigned int val; - int i; - int ret; - - lzo_blocks = codec->reg_cache; - for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { - WARN_ON(!snd_soc_codec_writable_register(codec, i)); - ret = snd_soc_cache_read(codec, i, &val); - if (ret) - return ret; - codec->cache_bypass = 1; - ret = snd_soc_write(codec, i, val); - codec->cache_bypass = 0; - if (ret) - return ret; - dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", - i, val); - } - - return 0; -} - -static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; - int ret, blkindex, blkpos; - size_t blksize, tmp_dst_len; - void *tmp_dst; - - /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); - /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); - /* size of the compressed block */ - blksize = snd_soc_lzo_get_blksize(codec); - lzo_blocks = codec->reg_cache; - lzo_block = lzo_blocks[blkindex]; - - /* save the pointer and length of the compressed block */ - tmp_dst = lzo_block->dst; - tmp_dst_len = lzo_block->dst_len; - - /* prepare the source to be the compressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* decompress the block */ - ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret < 0) { - kfree(lzo_block->dst); - goto out; - } - - /* write the new value to the cache */ - if (snd_soc_set_cache_val(lzo_block->dst, blkpos, value, - codec->driver->reg_word_size)) { - kfree(lzo_block->dst); - goto out; - } - - /* prepare the source to be the decompressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* compress the block */ - ret = snd_soc_lzo_compress_cache_block(codec, lzo_block); - if (ret < 0) { - kfree(lzo_block->dst); - kfree(lzo_block->src); - goto out; - } - - /* set the bit so we know we have to sync this register */ - set_bit(reg, lzo_block->sync_bmp); - kfree(tmp_dst); - kfree(lzo_block->src); - return 0; -out: - lzo_block->dst = tmp_dst; - lzo_block->dst_len = tmp_dst_len; - return ret; -} - -static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value) -{ - struct snd_soc_lzo_ctx *lzo_block, **lzo_blocks; - int ret, blkindex, blkpos; - size_t blksize, tmp_dst_len; - void *tmp_dst; - - *value = 0; - /* index of the compressed lzo block */ - blkindex = snd_soc_lzo_get_blkindex(codec, reg); - /* register index within the decompressed block */ - blkpos = snd_soc_lzo_get_blkpos(codec, reg); - /* size of the compressed block */ - blksize = snd_soc_lzo_get_blksize(codec); - lzo_blocks = codec->reg_cache; - lzo_block = lzo_blocks[blkindex]; - - /* save the pointer and length of the compressed block */ - tmp_dst = lzo_block->dst; - tmp_dst_len = lzo_block->dst_len; - - /* prepare the source to be the compressed block */ - lzo_block->src = lzo_block->dst; - lzo_block->src_len = lzo_block->dst_len; - - /* decompress the block */ - ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret >= 0) - /* fetch the value from the cache */ - *value = snd_soc_get_cache_val(lzo_block->dst, blkpos, - codec->driver->reg_word_size); - - kfree(lzo_block->dst); - /* restore the pointer and length of the compressed block */ - lzo_block->dst = tmp_dst; - lzo_block->dst_len = tmp_dst_len; - return 0; -} - -static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - int i, blkcount; - - lzo_blocks = codec->reg_cache; - if (!lzo_blocks) - return 0; - - blkcount = snd_soc_lzo_block_count(); - /* - * the pointer to the bitmap used for syncing the cache - * is shared amongst all lzo_blocks. Ensure it is freed - * only once. - */ - if (lzo_blocks[0]) - kfree(lzo_blocks[0]->sync_bmp); - for (i = 0; i < blkcount; ++i) { - if (lzo_blocks[i]) { - kfree(lzo_blocks[i]->wmem); - kfree(lzo_blocks[i]->dst); - } - /* each lzo_block is a pointer returned by kmalloc or NULL */ - kfree(lzo_blocks[i]); - } - kfree(lzo_blocks); - codec->reg_cache = NULL; - return 0; -} - -static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) -{ - struct snd_soc_lzo_ctx **lzo_blocks; - size_t bmp_size; - const struct snd_soc_codec_driver *codec_drv; - int ret, tofree, i, blksize, blkcount; - const char *p, *end; - unsigned long *sync_bmp; - - ret = 0; - codec_drv = codec->driver; - - /* - * If we have not been given a default register cache - * then allocate a dummy zero-ed out region, compress it - * and remember to free it afterwards. - */ - tofree = 0; - if (!codec->reg_def_copy) - tofree = 1; - - if (!codec->reg_def_copy) { - codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL); - if (!codec->reg_def_copy) - return -ENOMEM; - } - - blkcount = snd_soc_lzo_block_count(); - codec->reg_cache = kzalloc(blkcount * sizeof *lzo_blocks, - GFP_KERNEL); - if (!codec->reg_cache) { - ret = -ENOMEM; - goto err_tofree; - } - lzo_blocks = codec->reg_cache; - - /* - * allocate a bitmap to be used when syncing the cache with - * the hardware. Each time a register is modified, the corresponding - * bit is set in the bitmap, so we know that we have to sync - * that register. - */ - bmp_size = codec_drv->reg_cache_size; - sync_bmp = kmalloc(BITS_TO_LONGS(bmp_size) * sizeof(long), - GFP_KERNEL); - if (!sync_bmp) { - ret = -ENOMEM; - goto err; - } - bitmap_zero(sync_bmp, bmp_size); - - /* allocate the lzo blocks and initialize them */ - for (i = 0; i < blkcount; ++i) { - lzo_blocks[i] = kzalloc(sizeof **lzo_blocks, - GFP_KERNEL); - if (!lzo_blocks[i]) { - kfree(sync_bmp); - ret = -ENOMEM; - goto err; - } - lzo_blocks[i]->sync_bmp = sync_bmp; - lzo_blocks[i]->sync_bmp_nbits = bmp_size; - /* alloc the working space for the compressed block */ - ret = snd_soc_lzo_prepare(lzo_blocks[i]); - if (ret < 0) - goto err; - } - - blksize = snd_soc_lzo_get_blksize(codec); - p = codec->reg_def_copy; - end = codec->reg_def_copy + codec->reg_size; - /* compress the register map and fill the lzo blocks */ - for (i = 0; i < blkcount; ++i, p += blksize) { - lzo_blocks[i]->src = p; - if (p + blksize > end) - lzo_blocks[i]->src_len = end - p; - else - lzo_blocks[i]->src_len = blksize; - ret = snd_soc_lzo_compress_cache_block(codec, - lzo_blocks[i]); - if (ret < 0) - goto err; - lzo_blocks[i]->decompressed_size = - lzo_blocks[i]->src_len; - } - - if (tofree) { - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - } - return 0; -err: - snd_soc_cache_exit(codec); -err_tofree: - if (tofree) { - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - } - return ret; -} -#endif - static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) { int i; @@ -889,17 +516,6 @@ static const struct snd_soc_cache_ops cache_types[] = { .write = snd_soc_flat_cache_write, .sync = snd_soc_flat_cache_sync }, -#ifdef CONFIG_SND_SOC_CACHE_LZO - { - .id = SND_SOC_LZO_COMPRESSION, - .name = "LZO", - .init = snd_soc_lzo_cache_init, - .exit = snd_soc_lzo_cache_exit, - .read = snd_soc_lzo_cache_read, - .write = snd_soc_lzo_cache_write, - .sync = snd_soc_lzo_cache_sync - }, -#endif { .id = SND_SOC_RBTREE_COMPRESSION, .name = "rbtree", -- 1.7.7.2

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[alsa-devel] [PATCH v4] ASoC: Add new Realtek ALC5632 CODEC driver.
by leon＠leon.nu 10 Nov '11

10 Nov '11

From: Leon Romanovsky <leon(a)leon.nu> This driver implements basic functionality, using I²C for the control channel. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> --- v4: * Use dev_info instead dev_err for information messages. * Use EINVAL for incorrect pll_id. * Mark cache as needed to sync on resume. * Use mask for power registers to protect reserved bits. * Move power management logic from probe to set_bias function. * Remove references to sound platform device. v3: * DAC is now handled by DAPM * Power down control correct handling. * Fixed speaker dapm routes. * Remove unused variables. * Sync stream names in DAPM and codec dai. * Use update bits instead write to reg in power depop function. * Rename Route to AB-D Amp Mux * Add snd controls to codec driver. Don't register it in probe. * Add dapm widgets and routes to codec driver. Don't register it. * Don't force power for spk apm, main i2s. It is described in DAPM. * Set symmetric_rates flag to prevent different rates for playback and capture. * Removed commented DEBUG definition. * Use SOC infrastructure sync cache mechanism. * Remove non possible IF switch. * Fix indentation issues * Convert alc5632 MICBIAS to a supply widget * Remove codec input parameter * Remove VERSION define * Remove unnecessary dev_kfree() and update copyright notice * Control speaker amplifier with DAPM. * Let machine driver choose master/slave mode for the codec. * Clean up. removed dev_dbg. * Use devm_kzalloc/devm_kzfree for memory allocation/deallocation. * Fixed stereo headphones playback. * Set up correct name for Stereo DAC volume item. v2: Free from checkpatch warnings v1: Initial code drop --- --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/alc5632.c | 1153 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/alc5632.h | 249 ++++++++++ 4 files changed, 1407 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/alc5632.c create mode 100644 sound/soc/codecs/alc5632.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..684cc15 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C + select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -168,6 +169,8 @@ config SND_SOC_AK4671 config SND_SOC_ALC5623 tristate +config SND_SOC_ALC5632 + tristate config SND_SOC_CQ0093VC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..2ed5790 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o +snd-soc-alc5632-objs := alc5632.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -113,6 +114,7 @@ obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c new file mode 100644 index 0000000..68465d8 --- /dev/null +++ b/sound/soc/codecs/alc5632.c @@ -0,0 +1,1153 @@ +/* +* alc5632.c -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.c by Arnaud Patard +* +* 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/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/initval.h> + +#include "alc5632.h" + +/* + * ALC5632 register cache + */ +static const u16 alc5632_reg_defaults[] = { + 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ + 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ + 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ + 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ + 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ + 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ + 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ + 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ + 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ + 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ + 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ + 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ + 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ + 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ + 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ + 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ + 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +}; + +/* codec private data */ +struct alc5632_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; +}; + +static int alc5632_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ALC5632_RESET: + case ALC5632_PWR_DOWN_CTRL_STATUS: + case ALC5632_GPIO_PIN_STATUS: + case ALC5632_OVER_CURR_STATUS: + case ALC5632_HID_CTRL_DATA: + case ALC5632_EQ_CTRL: + return 1; + + default: + break; + } + + return 0; +} + +static inline int alc5632_reset(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ALC5632_RESET, 0); + return snd_soc_read(codec, ALC5632_RESET); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5632 Controls + */ + +/* -34.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +/* -46.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +/* -16.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +/* 0db min scale, 6 db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); +/* 0db min scalem 0.75db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); + +static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { + /* left starts at bit 8, right at bit 0 */ + /* 31 steps (5 bit), -46.5db scale */ + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + /* bit 15 mutes left, bit 7 right */ + SOC_DOUBLE("Line Playback Switch", + ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5632_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Voice DAC Playback Volume", + ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE_TLV("Phone Capture Volume", + ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Stereo DAC Playback Volume", + ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), + SOC_DOUBLE("Stereo DAC Playback Switch", + ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1), +SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1), +SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1), +SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 9, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1), +SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 10, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1), +SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5632_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Mute"}; +static const char *alc5632_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5632_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5632_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5632_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5632_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static const struct snd_kcontrol_new alc5632_auxout_mux_controls = +SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5632_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static const struct snd_kcontrol_new alc5632_spkout_mux_controls = +SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5632_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = +SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5632_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = +SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5632_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = +SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); + +/* speaker amplifier */ +static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5632_amp_enum = + SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static const struct snd_kcontrol_new alc5632_amp_mux_controls = + SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); + + +static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5632_hp_mixer_controls[0], + ARRAY_SIZE(alc5632_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0, + &alc5632_hpr_mixer_controls[0], + ARRAY_SIZE(alc5632_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0, + &alc5632_hpl_mixer_controls[0], + ARRAY_SIZE(alc5632_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, + &alc5632_mono_mixer_controls[0], + ARRAY_SIZE(alc5632_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, + &alc5632_speaker_mixer_controls[0], + ARRAY_SIZE(alc5632_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, + &alc5632_captureL_mixer_controls[0], + ARRAY_SIZE(alc5632_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, + &alc5632_captureR_mixer_controls[0], + ARRAY_SIZE(alc5632_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0), +SND_SOC_DAPM_MIXER("DAC Right Channel", + ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", ALC5632_PWR_MANAG_ADD1, 10, 0, + &alc5632_amp_mux_controls), + +SND_SOC_DAPM_OUTPUT("AUXOUT"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("PHONEP"), +SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + + +static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"Phone Mix", NULL, "Phone"}, + {"Phone Mix", NULL, "Phone ADMix"}, + {"AUXOUT", NULL, "Aux Out"}, + + /* DAC */ + {"DAC Right Channel", NULL, "I2S Mix"}, + {"DAC Left Channel", NULL, "I2S Mix"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + + {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, + {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, + + + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Aux Out", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Phone", NULL, "PHONEP"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"SpeakerOut N Mux", "RN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Right Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"Left Speaker", NULL, "AB-D Amp Mux"}, + {"Right Speaker", NULL, "AB-D Amp Mux"}, + + {"SPKOUT", NULL, "Left Speaker"}, + {"SPKOUT", NULL, "Right Speaker"}, + + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, + +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5632 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +/* FOUT = MCLK*(N+2)/((M+2)*(K+2)) + N: bit 15:8 (div 2 .. div 257) + K: bit 6:4 typical 2 + M: bit 3:0 (div 2 .. div 17) + + same as for 5623 - thanks! +*/ + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5632_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK) + return -EINVAL; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5632_DAI_CONTROL); + if (reg & ALC5632_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5632_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_BCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5632_PLL_FR_BCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_VBCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from voice clock */ + gbl_clk = ALC5632_PLL_FR_VBCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + + /* choose MCLK/BCLK/VBCLK */ + snd_soc_write(codec, ALC5632_GPCR2, gbl_clk); + /* choose PLL1 clock rate */ + snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div); + /* enable PLL1 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + ALC5632_PWR_ADD2_PLL1); + /* enable PLL2 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + ALC5632_PWR_ADD2_PLL2); + /* use PLL1 as main SYSCLK */ + snd_soc_update_bits(codec, ALC5632_GPCR1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {512*1, 0x3075}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5632->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5632->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5632_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5632_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5632_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5632_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5632_DAI_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5632_DAI_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); +} + +static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5632_DAI_CONTROL); + iface &= ~ALC5632_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5632_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5632_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5632_DAI_I2S_DL_24; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff); + + return 0; +} + +static int alc5632_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L \ + |ALC5632_MISC_HP_DEPOP_MUTE_R; + u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg); +} + +#define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF) + +#define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5632_ADD1_POWER_EN \ + (ALC5632_PWR_ADD1_DAC_REF \ + | ALC5632_PWR_ADD1_SOFTGEN_EN \ + | ALC5632_PWR_ADD1_HP_OUT_AMP \ + | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \ + | ALC5632_PWR_ADD1_MAIN_BIAS) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_SOFTGEN_EN, + ALC5632_PWR_ADD1_SOFTGEN_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, + ALC5632_ADD3_POWER_EN, + ALC5632_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + ALC5632_MISC_HP_DEPOP_MODE2_EN); + + /* "normal" mode: 0 @ 26 */ + /* set all PR0-7 mixers to 0 */ + snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, + ALC5632_PWR_DOWN_CTRL_STATUS_MASK, + 0); + + msleep(500); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_ADD2_POWER_EN, + ALC5632_ADD2_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_ADD1_POWER_EN, + ALC5632_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5632_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_MANAG_ADD1_MASK, + ALC5632_PWR_ADD1_MAIN_BIAS); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_MANAG_ADD2_MASK, + ALC5632_PWR_ADD2_VREF); + /* "normal" mode: 0 @ 26 */ + snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, + ALC5632_PWR_DOWN_CTRL_STATUS_MASK, + 0xffff ^ (ALC5632_PWR_VREF_PR3 + | ALC5632_PWR_VREF_PR2)); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_MANAG_ADD2_MASK, 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, + ALC5632_PWR_MANAG_ADD3_MASK, 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_MANAG_ADD1_MASK, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5632_dai_ops = { + .hw_params = alc5632_pcm_hw_params, + .digital_mute = alc5632_mute, + .set_fmt = alc5632_set_dai_fmt, + .set_sysclk = alc5632_set_dai_sysclk, + .set_pll = alc5632_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5632_dai = { + .name = "alc5632-hifi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + + .ops = &alc5632_dai_ops, + .symmetric_rates = 1, +}; + +static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5632_resume(struct snd_soc_codec *codec) +{ + int ret; + + /* mark cache as needed to sync */ + codec->cache_sync = 1; + + ret = snd_soc_cache_sync(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ + | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ + | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ + | ALC5632_ADC_REC_MONOMIX) + +#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ + | ALC5632_MIC_ROUTE_SPK \ + | ALC5632_MIC_ROUTE_MONOMIX) + +#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ + | ALC5632_PWR_DAC_STATUS \ + | ALC5632_PWR_AMIX_STATUS \ + | ALC5632_PWR_VREF_STATUS) + +#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ + / ALC5632_ADC_REC_GAIN_STEP) + +#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) + +#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) + +static int alc5632_probe(struct snd_soc_codec *codec) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5632_reset(codec); + + /* power on device */ + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + switch (alc5632->id) { + case 0x5c: + snd_soc_add_controls(codec, alc5632_vol_snd_controls, + ARRAY_SIZE(alc5632_vol_snd_controls)); + break; + default: + return -EINVAL; + } + + return ret; +} + +/* power down chip */ +static int alc5632_remove(struct snd_soc_codec *codec) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5632 = { + .probe = alc5632_probe, + .remove = alc5632_remove, + .suspend = alc5632_suspend, + .resume = alc5632_resume, + .set_bias_level = alc5632_set_bias_level, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = alc5632_reg_defaults, + .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_register = alc5632_volatile_register, + .controls = alc5632_snd_controls, + .num_controls = ARRAY_SIZE(alc5632_snd_controls), + .dapm_widgets = alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(alc5632_dapm_widgets), + .dapm_routes = alc5632_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), +}; + +/* + * alc5632 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5632_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5632_priv *alc5632; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid1: %x\n", vid1); + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_info(&client->dev, "got vid2: %x\n", vid2); + } + vid2 = (vid2 & 0xff); + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + alc5632 = devm_kzalloc(&client->dev, + sizeof(struct alc5632_priv), GFP_KERNEL); + if (alc5632 == NULL) + return -ENOMEM; + + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + return -EINVAL; + } + + i2c_set_clientdata(client, alc5632); + alc5632->control_data = client; + alc5632->control_type = SND_SOC_I2C; + mutex_init(&alc5632->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5632, &alc5632_dai, 1); + if (ret != 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int alc5632_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id alc5632_i2c_table[] = { + {"alc5632", 0x5c}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5632_i2c_driver = { + .driver = { + .name = "alc5632", + .owner = THIS_MODULE, + }, + .probe = alc5632_i2c_probe, + .remove = __devexit_p(alc5632_i2c_remove), + .id_table = alc5632_i2c_table, +}; + +static int __init alc5632_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5632_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5632_modinit); + +static void __exit alc5632_modexit(void) +{ + i2c_del_driver(&alc5632_i2c_driver); +} +module_exit(alc5632_modexit); + +MODULE_DESCRIPTION("ASoC ALC5632 driver"); +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h new file mode 100644 index 0000000..ff4c0fd --- /dev/null +++ b/sound/soc/codecs/alc5632.h @@ -0,0 +1,249 @@ +/* +* alc5632.h -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.h by Arnaud Patard +* +* 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 _ALC5632_H +#define _ALC5632_H + +#define ALC5632_RESET 0x00 +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5632_SPK_OUT_VOL 0x02 /* spe out vol */ +#define ALC5632_SPK_OUT_VOL_STEP 1.5 +#define ALC5632_HP_OUT_VOL 0x04 /* hp out vol */ +#define ALC5632_AUX_OUT_VOL 0x06 /* aux out vol */ +#define ALC5632_PHONE_IN_VOL 0x08 /* phone in vol */ +#define ALC5632_LINE_IN_VOL 0x0A /* line in vol */ +#define ALC5632_STEREO_DAC_IN_VOL 0x0C /* stereo dac in vol */ +#define ALC5632_MIC_VOL 0x0E /* mic in vol */ +/* stero dac/mic routing */ +#define ALC5632_MIC_ROUTING_CTRL 0x10 +#define ALC5632_MIC_ROUTE_MONOMIX (1 << 0) +#define ALC5632_MIC_ROUTE_SPK (1 << 1) +#define ALC5632_MIC_ROUTE_HP (1 << 2) + +#define ALC5632_ADC_REC_GAIN 0x12 /* rec gain */ +#define ALC5632_ADC_REC_GAIN_RANGE 0x1F1F +#define ALC5632_ADC_REC_GAIN_BASE (-16.5) +#define ALC5632_ADC_REC_GAIN_STEP 1.5 + +#define ALC5632_ADC_REC_MIXER 0x14 /* mixer control */ +#define ALC5632_ADC_REC_MIC1 (1 << 6) +#define ALC5632_ADC_REC_MIC2 (1 << 5) +#define ALC5632_ADC_REC_LINE_IN (1 << 4) +#define ALC5632_ADC_REC_AUX (1 << 3) +#define ALC5632_ADC_REC_HP (1 << 2) +#define ALC5632_ADC_REC_SPK (1 << 1) +#define ALC5632_ADC_REC_MONOMIX (1 << 0) + +#define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */ +/* ALC5632_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */ +#define ALC5632_OUTPUT_MIXER_RP (1 << 14) +#define ALC5632_OUTPUT_MIXER_WEEK (1 << 12) +#define ALC5632_OUTPUT_MIXER_HP (1 << 10) +#define ALC5632_OUTPUT_MIXER_AUX_SPK (2 << 6) +#define ALC5632_OUTPUT_MIXER_AUX_HP_LR (1 << 6) +#define ALC5632_OUTPUT_MIXER_HP_R (1 << 8) +#define ALC5632_OUTPUT_MIXER_HP_L (1 << 9) + +#define ALC5632_MIC_CTRL 0x22 /* mic phone ctrl */ +#define ALC5632_MIC_BOOST_BYPASS 0 +#define ALC5632_MIC_BOOST_20DB 1 +#define ALC5632_MIC_BOOST_30DB 2 +#define ALC5632_MIC_BOOST_40DB 3 + +#define ALC5632_DIGI_BOOST_CTRL 0x24 /* digi mic / bost ctl */ +#define ALC5632_MIC_BOOST_RANGE 7 +#define ALC5632_MIC_BOOST_STEP 6 +#define ALC5632_PWR_DOWN_CTRL_STATUS 0x26 +#define ALC5632_PWR_DOWN_CTRL_STATUS_MASK 0xEF00 +#define ALC5632_PWR_VREF_PR3 (1 << 11) +#define ALC5632_PWR_VREF_PR2 (1 << 10) +#define ALC5632_PWR_VREF_STATUS (1 << 3) +#define ALC5632_PWR_AMIX_STATUS (1 << 2) +#define ALC5632_PWR_DAC_STATUS (1 << 1) +#define ALC5632_PWR_ADC_STATUS (1 << 0) +/* stereo/voice DAC / stereo adc func ctrl */ +#define ALC5632_DAC_FUNC_SELECT 0x2E + +/* Main serial data port ctrl (i2s) */ +#define ALC5632_DAI_CONTROL 0x34 + +#define ALC5632_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5632_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5632_DAI_SADLRCK_MODE (1 << 14) +/* 0:voice, 1:main */ +#define ALC5632_DAI_MAIN_I2S_SYSCLK_SEL (1 << 8) +#define ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_MAIN_I2S_LRCK_INV (1 << 6) +#define ALC5632_DAI_I2S_DL_MASK (3 << 2) +#define ALC5632_DAI_I2S_DL_8 (3 << 2) +#define ALC5632_DAI_I2S_DL_24 (2 << 2) +#define ALC5632_DAI_I2S_DL_20 (1 << 2) +#define ALC5632_DAI_I2S_DL_16 (0 << 2) +#define ALC5632_DAI_I2S_DF_MASK (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_B (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_A (2 << 0) +#define ALC5632_DAI_I2S_DF_LEFT (1 << 0) +#define ALC5632_DAI_I2S_DF_I2S (0 << 0) +/* extend serial data port control (VoDAC_i2c/pcm) */ +#define ALC5632_DAI_CONTROL2 0x36 +/* 0:gpio func, 1:voice pcm */ +#define ALC5632_DAI_VOICE_PCM_ENABLE (1 << 15) +/* 0:master, 1:slave */ +#define ALC5632_DAI_VOICE_MODE_SEL (1 << 14) +/* 0:disable, 1:enable */ +#define ALC5632_DAI_HPF_CLK_CTRL (1 << 13) +/* 0:main, 1:voice */ +#define ALC5632_DAI_VOICE_I2S_SYSCLK_SEL (1 << 8) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_VBCLK_SYSCLK_SEL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_I2S_LR_INV (1 << 6) +#define ALC5632_DAI_VOICE_DL_MASK (3 << 2) +#define ALC5632_DAI_VOICE_DL_16 (0 << 2) +#define ALC5632_DAI_VOICE_DL_20 (1 << 2) +#define ALC5632_DAI_VOICE_DL_24 (2 << 2) +#define ALC5632_DAI_VOICE_DL_8 (3 << 2) +#define ALC5632_DAI_VOICE_DF_MASK (3 << 0) +#define ALC5632_DAI_VOICE_DF_I2S (0 << 0) +#define ALC5632_DAI_VOICE_DF_LEFT (1 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_A (2 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_B (3 << 0) + +#define ALC5632_PWR_MANAG_ADD1 0x3A +#define ALC5632_PWR_MANAG_ADD1_MASK 0xEFFF +#define ALC5632_PWR_ADD1_DAC_L_EN (1 << 15) +#define ALC5632_PWR_ADD1_DAC_R_EN (1 << 14) +#define ALC5632_PWR_ADD1_ZERO_CROSS (1 << 13) +#define ALC5632_PWR_ADD1_MAIN_I2S_EN (1 << 11) +#define ALC5632_PWR_ADD1_SPK_AMP_EN (1 << 10) +#define ALC5632_PWR_ADD1_HP_OUT_AMP (1 << 9) +#define ALC5632_PWR_ADD1_HP_OUT_ENH_AMP (1 << 8) +#define ALC5632_PWR_ADD1_VOICE_DAC_MIX (1 << 7) +#define ALC5632_PWR_ADD1_SOFTGEN_EN (1 << 6) +#define ALC5632_PWR_ADD1_MIC1_SHORT_CURR (1 << 5) +#define ALC5632_PWR_ADD1_MIC2_SHORT_CURR (1 << 4) +#define ALC5632_PWR_ADD1_MIC1_EN (1 << 3) +#define ALC5632_PWR_ADD1_MIC2_EN (1 << 2) +#define ALC5632_PWR_ADD1_MAIN_BIAS (1 << 1) +#define ALC5632_PWR_ADD1_DAC_REF (1 << 0) + +#define ALC5632_PWR_MANAG_ADD2 0x3C +#define ALC5632_PWR_MANAG_ADD2_MASK 0x7FFF +#define ALC5632_PWR_ADD2_PLL1 (1 << 15) +#define ALC5632_PWR_ADD2_PLL2 (1 << 14) +#define ALC5632_PWR_ADD2_VREF (1 << 13) +#define ALC5632_PWR_ADD2_OVT_DET (1 << 12) +#define ALC5632_PWR_ADD2_VOICE_DAC (1 << 10) +#define ALC5632_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5632_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5632_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5632_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5632_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5632_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5632_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5632_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5632_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5632_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5632_PWR_MANAG_ADD3 0x3E +#define ALC5632_PWR_MANAG_ADD3_MASK 0x7CFF +#define ALC5632_PWR_ADD3_AUXOUT_VOL (1 << 14) +#define ALC5632_PWR_ADD3_SPK_L_OUT (1 << 13) +#define ALC5632_PWR_ADD3_SPK_R_OUT (1 << 12) +#define ALC5632_PWR_ADD3_HP_L_OUT_VOL (1 << 11) +#define ALC5632_PWR_ADD3_HP_R_OUT_VOL (1 << 10) +#define ALC5632_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5632_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5632_PWR_ADD3_AUXIN_VOL (1 << 5) +#define ALC5632_PWR_ADD3_AUXIN_MIX (1 << 4) +#define ALC5632_PWR_ADD3_MIC1_VOL (1 << 3) +#define ALC5632_PWR_ADD3_MIC2_VOL (1 << 2) +#define ALC5632_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5632_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5632_GPCR1 0x40 +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1 (1 << 15) +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_MCLK (0 << 15) +#define ALC5632_GPCR1_DAC_HI_FLT_EN (1 << 10) +#define ALC5632_GPCR1_SPK_AMP_CTRL (7 << 1) +#define ALC5632_GPCR1_VDD_100 (5 << 1) +#define ALC5632_GPCR1_VDD_125 (4 << 1) +#define ALC5632_GPCR1_VDD_150 (3 << 1) +#define ALC5632_GPCR1_VDD_175 (2 << 1) +#define ALC5632_GPCR1_VDD_200 (1 << 1) +#define ALC5632_GPCR1_VDD_225 (0 << 1) + +#define ALC5632_GPCR2 0x42 +#define ALC5632_GPCR2_PLL1_SOUR_SEL (3 << 12) +#define ALC5632_PLL_FR_MCLK (0 << 12) +#define ALC5632_PLL_FR_BCLK (2 << 12) +#define ALC5632_PLL_FR_VBCLK (3 << 12) +#define ALC5632_GPCR2_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5632_PLL1_CTRL 0x44 +#define ALC5632_PLL1_CTRL_N_VAL(n) (((n) & 0x0f) << 8) +#define ALC5632_PLL1_M_BYPASS (1 << 7) +#define ALC5632_PLL1_CTRL_K_VAL(k) (((k) & 0x07) << 4) +#define ALC5632_PLL1_CTRL_M_VAL(m) (((m) & 0x0f) << 0) + +#define ALC5632_PLL2_CTRL 0x46 +#define ALC5632_PLL2_EN (1 << 15) +#define ALC5632_PLL2_RATIO (0 << 15) + +#define ALC5632_GPIO_PIN_CONFIG 0x4C +#define ALC5632_GPIO_PIN_POLARITY 0x4E +#define ALC5632_GPIO_PIN_STICKY 0x50 +#define ALC5632_GPIO_PIN_WAKEUP 0x52 +#define ALC5632_GPIO_PIN_STATUS 0x54 +#define ALC5632_GPIO_PIN_SHARING 0x56 +#define ALC5632_OVER_CURR_STATUS 0x58 +#define ALC5632_SOFTVOL_CTRL 0x5A +#define ALC5632_GPIO_OUPUT_PIN_CTRL 0x5C + +#define ALC5632_MISC_CTRL 0x5E +#define ALC5632_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5632_MISC_AVC_TRGT_SEL (3 << 12) +#define ALC5632_MISC_AVC_TRGT_RIGHT (1 << 12) +#define ALC5632_MISC_AVC_TRGT_LEFT (2 << 12) +#define ALC5632_MISC_AVC_TRGT_BOTH (3 << 12) +#define ALC5632_MISC_HP_DEPOP_MODE1_EN (1 << 9) +#define ALC5632_MISC_HP_DEPOP_MODE2_EN (1 << 8) +#define ALC5632_MISC_HP_DEPOP_MUTE_L (1 << 7) +#define ALC5632_MISC_HP_DEPOP_MUTE_R (1 << 6) +#define ALC5632_MISC_HP_DEPOP_MUTE (1 << 5) +#define ALC5632_MISC_GPIO_WAKEUP_CTRL (1 << 1) +#define ALC5632_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5632_DAC_CLK_CTRL1 0x60 +#define ALC5632_DAC_CLK_CTRL2 0x62 +#define ALC5632_DAC_CLK_CTRL2_DIV1_2 (1 << 0) +#define ALC5632_VOICE_DAC_PCM_CLK_CTRL1 0x64 +#define ALC5632_PSEUDO_SPATIAL_CTRL 0x68 +#define ALC5632_HID_CTRL_INDEX 0x6A +#define ALC5632_HID_CTRL_DATA 0x6C +#define ALC5632_EQ_CTRL 0x6E + +/* undocumented */ +#define ALC5632_VENDOR_ID1 0x7C +#define ALC5632_VENDOR_ID2 0x7E + +#endif -- 1.7.3.4

3 3

[alsa-devel] [PATCH] MIPS/sound: DB1200 audio support depends on MIPS arch.
by Manuel Lauss 10 Nov '11

10 Nov '11

DB1200 audio support should only be visible if MIPS/Alchemy arch is selected. Fixes an x86-allmodconfig build failure in linux-next. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index c620cbf9..a561040 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -52,7 +52,7 @@ config SND_SOC_DB1000 config SND_SOC_DB1200 tristate "DB1200/DB1300/DB1550 Audio support" - select SND_SOC_AU1XPSC + depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC select SND_SOC_WM9712 -- 1.7.7.2 -- 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

3 4

[alsa-devel] [PATCH] MAINTAINERS: Drop inactive Samsung ASoC maintainer
by Jassi Brar 10 Nov '11

10 Nov '11

Signed-off-by: Jassi Brar <jassisinghbrar(a)gmail.com> --- MAINTAINERS | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 2014c1f..c1a96f9 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5581,7 +5581,6 @@ F: drivers/media/video/*7146* F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS -M: Jassi Brar <jassisinghbrar(a)gmail.com> M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported -- 1.7.4.1

2 1

Re: [alsa-devel] Linux 3.2-rc1
by Linus Torvalds 10 Nov '11

10 Nov '11

Hmm, I don't know what caused this to trigger, but I'm adding both the i915 people and the HDA people to the cc, and they can fight to the death about this in the HDMI Thunderdome. Guys: One.. Two.. Three.. FIGHT! Linus On Tue, Nov 8, 2011 at 6:55 AM, Nick Bowler <nbowler(a)elliptictech.com> wrote: > > Mode switches are very noisy on an Intel G45 in 3.2-rc1: > > HDMI hot plug event: Codec=3 Pin=3 Presence_Detect=1 ELD_Valid=1 > HDMI status: Codec=3 Pin=3 Presence_Detect=1 ELD_Valid=1 > HDMI: detected monitor W2253 at connection type HDMI > HDMI: available speakers: FL/FR > HDMI: supports coding type LPCM: channels = 2, rates = 32000 44100 48000, bits = 16 20 24 > > These lines get printed every single switch; previously only a single > line was printed once at boot (the "HDMI status" line). > > Cheers, > -- > Nick Bowler, Elliptic Technologies (http://www.elliptictech.com/) >

3 3

[alsa-devel] [PATCH] ASoC: Remove extra space in runtime struct definition
by Mark Brown 10 Nov '11

10 Nov '11

My usual technique for finding definitions is to search for "name {" which breaks with the extra space. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 877fcc1..02a5c55 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -843,7 +843,7 @@ struct snd_soc_card { }; /* SoC machine DAI configuration, glues a codec and cpu DAI together */ -struct snd_soc_pcm_runtime { +struct snd_soc_pcm_runtime { struct device dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; -- 1.7.7.2

1 0

Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Mark Brown 10 Nov '11

10 Nov '11

On Thu, Nov 10, 2011 at 02:00:22PM +0900, Tomoya MORINAGA wrote: > (2011/11/08 23:38), Mark Brown wrote: > >On Tue, Nov 08, 2011 at 06:03:53PM +0900, Tomoya MORINAGA wrote: > >>struct snd_ml7213i2s_pcm { > >> enum snd_soc_control_type control_type; > >> struct snd_ml7213i2s *ml7213i2s; > >> spinlock_t lock; > >> unsigned int irq_pos; > >> unsigned int buf_pos; > >> struct snd_pcm_substream *substream; > >> struct cbdata cbd; /* i2s callback info */ > >> unsigned int channels; > >> unsigned int rw; > >> unsigned int rate; > >> unsigned int ch; > >> unsigned int setup_flag; > >> unsigned int format; > >> unsigned int bclkfs; > >> struct mutex i2c_mutex; > >>}; > >...this looks *really* confused, there's things in here which are a mix > >of DMA controller and CODEC driver things. The CODEC and DMA drivers > >shouldn't know anything about each other, let alone be referencing the > >same data structure. > Let me clarify your saying. > I couldn't understand what your "DMA controller" mean. > Which "snd_ml7213i2s_pcm" structure member do you mean ? Things like irq_pos and buf_pos for example.

1 0

[alsa-devel] [PATCH 0/3] ALSA: intel8x0: Improve virtual environment detection
by Konstantin Ozerkov 10 Nov '11

10 Nov '11

Improve detection logic as discussed early: add secondary checks by PCI SSID. Note: patch [3/3] is made for alsa-driver tree.

2 5

[alsa-devel] [PATCH v6] ASoC:Add support for cs42l73 codec
by Brian Austin 10 Nov '11

10 Nov '11

Signed-off-by:Brian Austin <brian.austin(a)cirrus.com> Signed-off-by:Georgi Vlaev <joe(a)nucleusys.com> This patch adds support for Cirrus Logic CS42L73 low power stereo codec --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1255 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 ++++++++ 4 files changed, 1488 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..4f0ce61 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -175,6 +176,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a7c415d..a76475d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..e191f44 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1255 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com> + * Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.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 "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + u32 sysclk; + u8 mclksel; + u32 mclk; + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[] = { +/*0*/ 0x00, 0x42, 0xA7, 0x30, +/*4*/ 0x00, 0x00, 0xF1, 0xDF, +/*8*/ 0x3F, 0x57, 0x53, 0x00, +/*C*/ 0x00, 0x15, 0x00, 0x15, +/*A*/ 0x00, 0x15, 0x00, 0x06, +/*E*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +}; + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const struct snd_kcontrol_new input_left_mixer[] = { + SOC_DAPM_SINGLE("ADC Left Input", CS42L73_PWRCTL1, + 5, 1, 1), + SOC_DAPM_SINGLE("DMIC Left Input", CS42L73_PWRCTL1, + 4, 1, 1), +}; + +static const struct snd_kcontrol_new input_right_mixer[] = { + SOC_DAPM_SINGLE("ADC Right Input", CS42L73_PWRCTL1, + 7, 1, 1), + SOC_DAPM_SINGLE("DMIC Right Input", CS42L73_PWRCTL1, + 6, 1, 1), +}; + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mix_texts[] = { + "Left", "Right", "Mono Mix"}; + +static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; + +static const struct soc_enum spk_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_asp_mixer = + SOC_DAPM_ENUM("Route", spk_asp_enum); + +static const struct soc_enum spk_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 4, 3, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new spk_xsp_mixer = + SOC_DAPM_ENUM("Route", spk_xsp_enum); + +static const struct soc_enum esl_asp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_asp_mixer = + SOC_DAPM_ENUM("Route", esl_asp_enum); + +static const struct soc_enum esl_xsp_enum = + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + ARRAY_SIZE(cs42l73_mono_mix_texts), + cs42l73_mono_mix_texts, + cs42l73_mono_mix_values); + +static const struct snd_kcontrol_new esl_xsp_mixer = + SOC_DAPM_ENUM("Route", esl_xsp_enum); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xsp_output_mux_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new vsp_output_mux = + SOC_DAPM_ENUM("VSPOUT Mux", vsp_output_mux_enum); + +static const struct snd_kcontrol_new xsp_output_mux = + SOC_DAPM_ENUM("XSPOUT Mux", xsp_output_mux_enum); + +static const struct snd_kcontrol_new hp_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new lo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1); + +static const struct snd_kcontrol_new spk_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new spklo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new ear_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, + 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, + 6, 1, 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 0, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 0, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Left Mux", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Right Mux", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("DMIC Left", NULL, CS42L73_PWRCTL1, 6, 1), + SND_SOC_DAPM_ADC("DMIC Right", NULL, CS42L73_PWRCTL1, 4, 1), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Left Capture", SND_SOC_NOPM, + 0, 0, input_left_mixer, + ARRAY_SIZE(input_left_mixer)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Right Capture", SND_SOC_NOPM, + 0, 0, input_right_mixer, + ARRAY_SIZE(input_right_mixer)), + + SND_SOC_DAPM_MIXER("ASPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("VSPOUT Mux", SND_SOC_NOPM, + 0, 0, &vsp_output_mux), + + SND_SOC_DAPM_MUX("XSPOUT Mux", SND_SOC_NOPM, + 0, 0, &xsp_output_mux), + + SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + + SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("HL Right Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("SPK Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("ESL Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("ESL-XSP Mux", SND_SOC_NOPM, + 0, 0, &esl_xsp_mixer), + + SND_SOC_DAPM_MUX("ESL-ASP Mux", SND_SOC_NOPM, + 0, 0, &esl_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-ASP Mux", SND_SOC_NOPM, + 0, 0, &spk_asp_mixer), + + SND_SOC_DAPM_MUX("SPK-XSP Mux", SND_SOC_NOPM, + 0, 0, &spk_xsp_mixer), + + SND_SOC_DAPM_PGA("HL Left DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HL Right DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl), + SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, + &lo_amp_ctl), + SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl), + SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl), + SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("LINEOUTA"), + SND_SOC_DAPM_OUTPUT("LINEOUTB"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + + /* SPKLO EARSPK Paths */ + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"EAR Amp", "Switch", "ESL DAC"}, + {"SPKLO Amp", "Switch", "ESL DAC"}, + + {"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"}, + {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"}, + {"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"}, + + {"ESL Mixer", NULL, "ESL-ASP Mux"}, + {"ESL Mixer", NULL, "ESL-XSP Mux"}, + + {"ESL-ASP Mux", "Left", "ASPINL"}, + {"ESL-ASP Mux", "Right", "ASPINR"}, + {"ESL-ASP Mux", "Mono Mix", "ASPINM"}, + + {"ESL-XSP Mux", "Left", "XSPINL"}, + {"ESL-XSP Mux", "Right", "XSPINR"}, + {"ESL-XSP Mux", "Mono Mix", "XSPINM"}, + + /* Speakerphone Paths */ + {"SPKOUT", NULL, "SPK Amp"}, + {"SPK Amp", "Switch", "SPK DAC"}, + + {"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"}, + {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"}, + /* Loopback */ + {"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"}, + {"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"}, + + {"SPK Mixer", NULL, "SPK-ASP Mux"}, + {"SPK Mixer", NULL, "SPK-XSP Mux"}, + + {"SPK-ASP Mux", "Left", "ASPINL"}, + {"SPK-ASP Mux", "Mono Mix", "ASPINM"}, + {"SPK-ASP Mux", "Right", "ASPINR"}, + + {"SPK-XSP Mux", "Left", "XSPINL"}, + {"SPK-XSP Mux", "Mono Mix", "XSPINM"}, + {"SPK-XSP Mux", "Right", "XSPINR"}, + + /* HP LineOUT Paths */ + {"HPOUTA", NULL, "HP Amp"}, + {"HPOUTB", NULL, "HP Amp"}, + {"LINEOUTA", NULL, "LO Amp"}, + {"LINEOUTB", NULL, "LO Amp"}, + + {"HP Amp", "Switch", "HL Left DAC"}, + {"HP Amp", "Switch", "HL Right DAC"}, + {"LO Amp", "Switch", "HL Left DAC"}, + {"LO Amp", "Switch", "HL Right DAC"}, + + {"HL Left DAC", "HL-XSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-XSP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-ASP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-ASP Volume", "HL Right Mixer"}, + {"HL Left DAC", "HL-VSP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-VSP Volume", "HL Right Mixer"}, + /* Loopback */ + {"HL Left DAC", "HL-IP Volume", "HL Left Mixer"}, + {"HL Right DAC", "HL-IP Volume", "HL Right Mixer"}, + {"HL Left Mixer", NULL, "Input Left Capture"}, + {"HL Right Mixer", NULL, "Input Right Capture"}, + + {"HL Left Mixer", NULL, "ASPINL"}, + {"HL Right Mixer", NULL, "ASPINR"}, + {"HL Left Mixer", NULL, "XSPINL"}, + {"HL Right Mixer", NULL, "XSPINR"}, + {"HL Left Mixer", NULL, "VSPIN"}, + {"HL Right Mixer", NULL, "VSPIN"}, + + /* Capture Paths */ + {"MIC1", NULL, "MIC1 Bias"}, + {"PGA Left Mux", "Mic 1", "MIC1"}, + {"MIC2", NULL, "MIC2 Bias"}, + {"PGA Right Mux", "Mic 2", "MIC2"}, + + {"PGA Left Mux", "Line A", "LINEINA"}, + {"PGA Right Mux", "Line B", "LINEINB"}, + + {"PGA Left", NULL, "PGA Left Mux"}, + {"PGA Right", NULL, "PGA Right Mux"}, + + {"ADC Left", NULL, "PGA Left"}, + {"ADC Right", NULL, "PGA Right"}, + + {"Input Left Capture", "ADC Left Input", "ADC Left"}, + {"Input Right Capture", "ADC Right Input", "ADC Right"}, + {"Input Left Capture", "DMIC Left Input", "DMIC Left"}, + {"Input Right Capture", "DMIC Right Input", "DMIC Right"}, + + /* Audio Capture */ + {"ASPL Output Mixer", NULL, "Input Left Capture"}, + {"ASPR Output Mixer", NULL, "Input Right Capture"}, + + {"ASPOUTL", "ASP-IP Volume", "ASPL Output Mixer"}, + {"ASPOUTR", "ASP-IP Volume", "ASPR Output Mixer"}, + + /* Auxillary Capture */ + {"XSPL Output Mixer", NULL, "Input Left Capture"}, + {"XSPR Output Mixer", NULL, "Input Right Capture"}, + + {"XSPOUTL", "XSP-IP Volume", "XSPL Output Mixer"}, + {"XSPOUTR", "XSP-IP Volume", "XSPR Output Mixer"}, + + {"XSPOUT Mux", NULL, "XSPL Output Mixer"}, + {"XSPOUT Mux", NULL, "XSPR Output Mixer"}, + + {"XSPOUTL", "Mono", "XSPOUT Mux"}, + {"XSPOUTR", "Mono", "XSPOUT Mux"}, + + {"XSPOUTL", "Stereo", "XSPOUT Mux"}, + {"XSPOUTR", "Stereo", "XSPOUT Mux"}, + + /* Voice Capture */ + {"VSPL Output Mixer", NULL, "Input Left Capture"}, + {"VSPR Output Mixer", NULL, "Input Left Capture"}, + + {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"}, + {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"}, + + {"VSPOUT Mux", NULL, "VSPL Output Mixer"}, + {"VSPOUT Mux", NULL, "VSPR Output Mixer"}, + + {"VSPOUTL", "Mono", "VSPOUT Mux"}, + {"VSPOUTR", "Mono", "VSPOUT Mux"}, + + {"VSPOUTL", "Stereo", "VSPOUT Mux"}, + {"VSPOUTR", "Stereo", "VSPOUT Mux"}, +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format in slave mode only\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } else { + /* CS42L73 Slave */ + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + + cs42l73_update_asrc(codec, id, srate); + + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), + 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = snd_soc_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = snd_soc_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = snd_soc_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = snd_soc_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = ARRAY_SIZE(cs42l73_reg), + .reg_cache_default = cs42l73_reg, + .reg_word_size = sizeof(u8), + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..188f06d --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Brian Austin <brian.austin(a)cirrus.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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- 1.7.7.1

3 3

[alsa-devel] [PATCH] ASoC: Fix duplicate const warnings in da7210.c
by Mark Brown 10 Nov '11

10 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/da7210.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index b545b7d..8b5848a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -240,7 +240,7 @@ static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ -static const char const *da7210_hpf_cutoff_txt[] = { +static const char * const da7210_hpf_cutoff_txt[] = { "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" }; @@ -251,7 +251,7 @@ static const struct soc_enum da7210_adc_hpf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); /* ADC and DAC voice (8kHz) high pass cutoff value */ -static const char const *da7210_vf_cutoff_txt[] = { +static const char * const da7210_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -- 1.7.7.2

1 0

[alsa-devel] Playback and capture synchronization
by Patrick Lai 10 Nov '11

10 Nov '11

Hi, I have a use case requiring playback and capture starting at the same time. Latency between starting of playback and capture streams need to be less than few PCM samples. Base on my understanding on ALSA which does not have the concept of grouping multiple streams, the best I can do is getting playback and capture to prepared state then trigger start two streams one after the other. Then, the latency would be the time for trigger start of first stream to complete. Is there a better way to deal with my use case from ALSA API standpoint? 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] ASoC: Convert WM5100 gpiolib support to direct regmap API usage
by Mark Brown 09 Nov '11

09 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 52 +++++++++++++++++++++----------------------- 1 files changed, 25 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 77ffde0..a00ef95 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2353,24 +2353,22 @@ static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip) static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; - snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, - WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); } static int wm5100_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; int val, ret; val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); - ret = snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, - WM5100_GP1_FN_MASK | WM5100_GP1_DIR | - WM5100_GP1_LVL, val); + ret = regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR | + WM5100_GP1_LVL, val); if (ret < 0) return ret; else @@ -2380,25 +2378,24 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip, static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; + unsigned int reg; int ret; - ret = snd_soc_read(codec, WM5100_GPIO_CTRL_1 + offset); + ret = regmap_read(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, &reg); if (ret < 0) return ret; - return (ret & WM5100_GP1_LVL) != 0; + return (reg & WM5100_GP1_LVL) != 0; } static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); - struct snd_soc_codec *codec = wm5100->codec; - return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, - WM5100_GP1_FN_MASK | WM5100_GP1_DIR, - (1 << WM5100_GP1_FN_SHIFT) | - (1 << WM5100_GP1_DIR_SHIFT)); + return regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR, + (1 << WM5100_GP1_FN_SHIFT) | + (1 << WM5100_GP1_DIR_SHIFT)); } static struct gpio_chip wm5100_template_chip = { @@ -2411,14 +2408,14 @@ static struct gpio_chip wm5100_template_chip = { .can_sleep = 1, }; -static void wm5100_init_gpio(struct snd_soc_codec *codec) +static void wm5100_init_gpio(struct i2c_client *i2c) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); int ret; wm5100->gpio_chip = wm5100_template_chip; wm5100->gpio_chip.ngpio = 6; - wm5100->gpio_chip.dev = codec->dev; + wm5100->gpio_chip.dev = &i2c->dev; if (wm5100->pdata.gpio_base) wm5100->gpio_chip.base = wm5100->pdata.gpio_base; @@ -2427,24 +2424,24 @@ static void wm5100_init_gpio(struct snd_soc_codec *codec) ret = gpiochip_add(&wm5100->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } -static void wm5100_free_gpio(struct snd_soc_codec *codec) +static void wm5100_free_gpio(struct i2c_client *i2c) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); int ret; ret = gpiochip_remove(&wm5100->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); } #else -static void wm5100_init_gpio(struct snd_soc_codec *codec) +static void wm5100_init_gpio(struct i2c_client *i2c) { } -static void wm5100_free_gpio(struct snd_soc_codec *codec) +static void wm5100_free_gpio(struct i2c_client *i2c) { } #endif @@ -2466,7 +2463,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) regcache_cache_only(wm5100->regmap, true); - wm5100_init_gpio(codec); for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, @@ -2574,7 +2570,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) err_gpio: if (i2c->irq) free_irq(i2c->irq, codec); - wm5100_free_gpio(codec); return ret; } @@ -2590,7 +2585,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) } if (i2c->irq) free_irq(i2c->irq, codec); - wm5100_free_gpio(codec); return 0; } @@ -2744,6 +2738,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, goto err_reset; } + wm5100_init_gpio(i2c); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); @@ -2755,6 +2751,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, return ret; err_reset: + wm5100_free_gpio(i2c); if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 1); gpio_free(wm5100->pdata.reset); @@ -2788,6 +2785,7 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *client) struct wm5100_priv *wm5100 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + wm5100_free_gpio(client); if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 1); gpio_free(wm5100->pdata.reset); -- 1.7.7.2

1 0

[alsa-devel] [PATCH] ASoC: Add Lowland machine driver
by Mark Brown 09 Nov '11

09 Nov '11

The Lowland platform is based on the Cragganmore system like Speyside but uses the WM5100 audio CODEC. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/Kconfig | 7 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/lowland.c | 246 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 255 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/lowland.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index c7598ed..4520cc1 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -199,3 +199,10 @@ config SND_SOC_SPEYSIDE_WM8962 depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 select SND_SAMSUNG_I2S select SND_SOC_WM8962 + +config SND_SOC_LOWLAND + tristate "Audio support for Wolfson Lowland" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM5100 + select SND_SOC_WM9081 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 8509d3c..7802c25 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -40,6 +40,7 @@ snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o snd-soc-speyside-wm8962-objs := speyside_wm8962.o +snd-soc-lowland-objs := lowland.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -61,3 +62,4 @@ obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o +obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c new file mode 100644 index 0000000..eff1b4b --- /dev/null +++ b/sound/soc/samsung/lowland.c @@ -0,0 +1,246 @@ +/* + * Lowland audio support + * + * Copyright 2011 Wolfson Microelectronics + * + * 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. + */ + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/gpio.h> +#include <linux/module.h> + +#include "../codecs/wm5100.h" +#include "../codecs/wm9081.h" + +#define MCLK1_RATE (44100 * 512) +#define CLKOUT_RATE (44100 * 256) + +static int lowland_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; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops lowland_ops = { + .hw_params = lowland_hw_params, +}; + +static struct snd_soc_jack lowland_headset; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin lowland_headset_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_codec_set_sysclk(codec, WM5100_CLK_SYSCLK, + WM5100_CLKSRC_MCLK1, MCLK1_RATE, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err("Failed to set SYSCLK clock source: %d\n", ret); + return ret; + } + + /* Clock OPCLK, used by the other audio components. */ + ret = snd_soc_codec_set_sysclk(codec, WM5100_CLK_OPCLK, 0, + CLKOUT_RATE, 0); + if (ret < 0) { + pr_err("Failed to set OPCLK rate: %d\n", ret); + return ret; + } + + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0, + &lowland_headset); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&lowland_headset, + ARRAY_SIZE(lowland_headset_pins), + lowland_headset_pins); + if (ret) + return ret; + + wm5100_detect(codec, &lowland_headset); + + return 0; +} + +static struct snd_soc_dai_link lowland_dai[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5100-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5100.1-001a", + .ops = &lowland_ops, + .init = lowland_wm5100_init, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5100-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .ops = &lowland_ops, + .ignore_suspend = 1, + }, +}; + +static int lowland_wm9081_init(struct snd_soc_dapm_context *dapm) +{ + snd_soc_dapm_nc_pin(dapm, "LINEOUT"); + + /* At any time the WM9081 is active it will have this clock */ + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, + CLKOUT_RATE, 0); +} + +static struct snd_soc_aux_dev lowland_aux_dev[] = { + { + .name = "wm9081", + .codec_name = "wm9081.1-006c", + .init = lowland_wm9081_init, + }, +}; + +static struct snd_soc_codec_conf lowland_codec_conf[] = { + { + .dev_name = "wm9081.1-006c", + .name_prefix = "Sub", + }, +}; + +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("Main Speaker"), + SOC_DAPM_PIN_SWITCH("Main DMIC"), + SOC_DAPM_PIN_SWITCH("Main AMIC"), + SOC_DAPM_PIN_SWITCH("WM1250 Input"), + SOC_DAPM_PIN_SWITCH("WM1250 Output"), + SOC_DAPM_PIN_SWITCH("Headphone"), +}; + +static struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_SPK("Main Speaker", NULL), + + SND_SOC_DAPM_MIC("Main AMIC", NULL), + SND_SOC_DAPM_MIC("Main DMIC", NULL), +}; + +static struct snd_soc_dapm_route audio_paths[] = { + { "Sub IN1", NULL, "HPOUT2L" }, + { "Sub IN2", NULL, "HPOUT2R" }, + + { "Main Speaker", NULL, "Sub SPKN" }, + { "Main Speaker", NULL, "Sub SPKP" }, + { "Main Speaker", NULL, "SPKDAT1" }, +}; + +static struct snd_soc_card lowland = { + .name = "Lowland", + .dai_link = lowland_dai, + .num_links = ARRAY_SIZE(lowland_dai), + .aux_dev = lowland_aux_dev, + .num_aux_devs = ARRAY_SIZE(lowland_aux_dev), + .codec_conf = lowland_codec_conf, + .num_configs = ARRAY_SIZE(lowland_codec_conf), + + .controls = controls, + .num_controls = ARRAY_SIZE(controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = audio_paths, + .num_dapm_routes = ARRAY_SIZE(audio_paths), +}; + +static __devinit int lowland_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &lowland; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } + + return 0; +} + +static int __devexit lowland_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver lowland_driver = { + .driver = { + .name = "lowland", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = lowland_probe, + .remove = __devexit_p(lowland_remove), +}; + +static int __init lowland_audio_init(void) +{ + return platform_driver_register(&lowland_driver); +} +module_init(lowland_audio_init); + +static void __exit lowland_audio_exit(void) +{ + platform_driver_unregister(&lowland_driver); +} +module_exit(lowland_audio_exit); + +MODULE_DESCRIPTION("Lowland audio support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:lowland"); -- 1.7.7.2

1 0

[alsa-devel] [PATCH 1/3] ASoC: Need to convert wm5100 cache sync to direct regmap usage too
by Mark Brown 09 Nov '11

09 Nov '11

ASoC knows nothing about the cache now. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index cf66209..01959e1 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1401,7 +1401,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, break; } - snd_soc_cache_sync(codec); + regcache_sync(wm5100->regmap); } break; -- 1.7.7.2

1 2

[alsa-devel] OOPS: Vmaster not freed on reconfig
by David Henningsson 09 Nov '11

09 Nov '11

Hi, I'm playing around with the reconfig functions of snd-hda-codec and got an OOPS in slave_put. I tried to research this, and I think I've nailed down the problem: the vmaster is not freed on reconfig (presumably, it should be fixed in hda_codec.c:snd_hda_ctls_clear). I'm a little unsure of whether the best thing is to just add it to codec.mixers, keep them in a separate array, or something else. What do you think? (It's on 3.0, it's a sigmatel/IDT, and I've got the screen captured by a camera if anyone is interested.) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 5

[alsa-devel] [PATCH v3 0/3] ARM: mxs: add recording support for saif
by Dong Aisheng 09 Nov '11

09 Nov '11

Changes since v2: * separate clkmux code into another patch according to Uwe * add lock according to Wolfram * Other minus fixes suggested by Uwe and Wolfram. * remove Wolfram's fix saif clock setting patch which is in v2 series since Wolfram will reform the clock code and send it out himself. For test purpose, user still needs that patch. People can get it from v2 series. The patch is: [PATCH v2 3/3] arm: mxs: disable clock-gates when setting saif-clocks The patches are based on imx-features branch since commit: f4f01e31835f. Changes since v1: The main changes are move mach-specific code(clkmux in DIGCTL) from saif driver to mach-specific layer based on Wolfram's suggestion. Note that the last patch is a RFC patch and sent out for testing since without that patch the saif may not work. Dong Aisheng (3): ARM: mxs: add saif clkmux functions ARM: mx28evk: add platform data for saif ARM: mx28evk: set a initial clock rate for saif arch/arm/mach-mxs/clock-mx28.c | 67 +++++++++++++++++++++++ arch/arm/mach-mxs/devices-mx28.h | 3 +- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 +- arch/arm/mach-mxs/include/mach/common.h | 2 + arch/arm/mach-mxs/include/mach/devices-common.h | 4 +- arch/arm/mach-mxs/include/mach/digctl.h | 21 +++++++ arch/arm/mach-mxs/mach-mx28evk.c | 16 +++++- 7 files changed, 112 insertions(+), 6 deletions(-) create mode 100644 arch/arm/mach-mxs/include/mach/digctl.h

4 7

[alsa-devel] [PATCH v4] ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
by Axel Lin 09 Nov '11

09 Nov '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9081.c | 81 +++++++++++++++++++------------------------- 1 files changed, 35 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b491ae1..f7c0738 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -826,84 +826,74 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | + WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP, 0); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source and disable bias */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - reg &= ~WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC); /* Disable VMID with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1291,11 +1281,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, -- 1.7.5.4

2 1

[alsa-devel] [PATCH] snd_usb_audio: add Logitech HD Webcam c510 to quirk-384
by Alexey Fisher 09 Nov '11

09 Nov '11

Logitech HD Webcam c510 provide wrong mixer resolution. Add it to "res = 384" quirk. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/usb/mixer.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c5444e0..ab23869 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -799,6 +799,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x0808): case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- 1.7.5.4

2 2

[alsa-devel] Fix internal mic on Dell Vostro 3500
by Julian Wollrath 09 Nov '11

09 Nov '11

Hello, the internal mic of my Dell Vostro 3500 was not working with kernel v3.2-rc1 (and propably with none of the older kernels). By choosing dell-s14 as the model I was able to get the internal mic working, but now my externel mic jack was a second headphone out jack. So I looked at the differences between the pin configurations of the auto model and the dell- s14 model and found out, that setting the value of on 0x11 from default 0x400000f6 to 0x90a60160 was responsible for the working internal mic. I attached a patch that fixes this problem but it has a flaw. Since I did not know what value I have to put in the second argument of SND_PCI_QUIRK I put 0x0000 as dummy in there. It works for me but if anyone could give me a hint on how to obtain, what I really have to put in there, I would be happy to fill in the right value. With best regards Julian Wollrath

2 3

[alsa-devel] [PATCH v3] ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
by Axel Lin 09 Nov '11

09 Nov '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9081.c | 82 ++++++++++++++++++++------------------------- 1 files changed, 36 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b491ae1..f032078 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -826,84 +826,75 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | + WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | + WM9081_VMID_SEL_MASK, 0); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source and disable bias */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - reg &= ~WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC); /* Disable VMID with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1291,11 +1282,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, -- 1.7.5.4

1 2

[alsa-devel] Spdif: Control interface
by ajay khandelwal 09 Nov '11

09 Nov '11

Hi All, On my PC SPDIF control is exposing mute and volume control. I understand for mute but how volume control is exported for digital audio. Best Regards, Ajay

3 5

Re: [alsa-devel] Why my linux OS has only 8 sound cards ? How can I modify it ?
by Takashi Iwai 09 Nov '11

09 Nov '11

[Please stop using HTML mail for posting to a ML...] At Wed, 9 Nov 2011 10:24:41 +0800, 沈发旺 wrote: > > Hi,Takashi Iwai : > * I used your patch ,but it also display this when I used the command > "ls /dev/dsp*".* > */dev/dsp /dev/dsp2 /dev/dsp4 /dev/dsp6 > /dev/dsp1 /dev/dsp3 /dev/dsp5 /dev/dsp7* > *It also display this when board up:* > *input: C-Media USB Headphone Set as > /devices/platform/fsl-ehci.1/usb2/2-1/2-1.5/2-1.5:1.3/input/input7 > generic-usb 0003:0D8C:000C.0007: input: USB HID v1.00 Device [C-Media USB > Headphone Set ] on usb-fsl-ehci.1-1.5/input3 > usb 2-1.6: new full speed USB device using fsl-ehci and address 8 > unable to register OSS PCM device 8:0 > unable to register OSS mixer device 8:0 > input: C-Media USB Headphone Set as > /devices/platform/fsl-ehci.1/usb2/2-1/2-1.6/2-1.6:1.3/input/input8 > generic-usb 0003:0D8C:000C.0008: input: USB HID v1.00 Device [C-Media USB > Headphone Set ] on usb-fsl-ehci.1-1.6/input3 > usb 2-1.7: new full speed USB device using fsl-ehci and address 9 > unable to register OSS PCM device 9:0 > unable to register OSS mixer device 9:0 > input: C-Media USB Headphone Set as > /devices/platform/fsl-ehci.1/usb2/2-1/2-1.7/2-1.7:1.3/input/input9 > generic-usb 0003:0D8C:000C.0009: input: USB HID v1.00 Device [C-Media USB > Headphone Set ] on usb-fsl-ehci.1-1.7/input3 > **when the sound cards exceed 8 ,I found it return err from > sound/sound_core.c:__sound_insert_unit( ) . I read this code and found > modify the function of "register_sound_special_device " -> max_unit = 128 + > chain; when I change to :** max_unit = 256 + chain, my board will display > 10 sound cards :* So it's a limitation in sound_core.c. (BTW, 256+chain is wrong. It should be simply 256.) I vaguely remember why we left the limit 8 in ALSA side. One reason was that sound_core itself doesn't support more than 8, and we weren't sure whether changing it would regress or not. For example, don't apps expect only a single digit after /dev/dsp? OTOH, having 11 sound devices with OSS must be a rare case and I'm sure a legacy system using OSS won't have such a configuration. And fixing the code is basically a right thing. That being said, I'm fine to change that part for the upstream, if no one objects. thanks, Takashi > /dev/dsp /dev/dsp2 /dev/dsp4 /dev/dsp6 /dev/dsp8 > /dev/dsp1 /dev/dsp3 /dev/dsp5 /dev/dsp7 /dev/dsp9 > *Thank you vary mach !* > 2011/11/9 Takashi Iwai <tiwai(a)suse.de> > > > At Tue, 08 Nov 2011 18:37:53 +0100, > > Clemens Ladisch wrote: > > > > > > Takashi Iwai wrote: > > > > Linus Torvalds wrote: > > > > > Takashi, any idea? Do we have some 8-sound-card limit somewhere? > > > > > > > > > > I do see that we have that odd > > > > > > > > > > if (card && card->number >= 8) > > > > > return 0; /* ignore silently */ > > > > > > > > > > in sound/core/sound_oss.c: snd_register_oss_device(), but it has no > > > > > comments. The thing goes back to 2005 with the commit log entry of > > > > > "[ALSA] dynamic minors (6/6): increase maximum number of sound > > cards", > > > > > but the value '8'isn't explained. > > > > > > > > ALSA itself can support 32 cards when CONFIG_SND_DYNAMIC_MINORS is > > > > set ... > > > > The above code is, however, about OSS-emulation. And I guess it's > > > > just a left-over of the old static limitation. > > > > > > Yes; I didn't change the limit in the OSS emulation because the minor > > > number layout doesn't have space for all 32 cards, and extending the > > > limit to 16 did not seem worth the effort for a 'legacy' interface. > > > > > > > Actually OSS could support up to 16 cards (in theory). > > > > > > > > Below is an untested fix patch. Does it work? > > > > > > Not without increasing SNDRV_OSS_MINORS, too: > > > > Ah, good catch. Below is the revised patch. > > > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: Fix the card number limit of OSS-emulation > > > > There are left-over codes from the ancient days with the static device > > number limitation of 8. Actaully OSS can support up to 16 cards. > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > v2: fix SNDRV_OSS_MINORS, too > > > > sound/core/sound_oss.c | 6 +++--- > > 1 files changed, 3 insertions(+), 3 deletions(-) > > > > diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c > > index c700920..e952833 100644 > > --- a/sound/core/sound_oss.c > > +++ b/sound/core/sound_oss.c > > @@ -35,7 +35,7 @@ > > #include <linux/sound.h> > > #include <linux/mutex.h> > > > > -#define SNDRV_OSS_MINORS 128 > > +#define SNDRV_OSS_MINORS 256 > > > > static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS]; > > static DEFINE_MUTEX(sound_oss_mutex); > > @@ -111,7 +111,7 @@ int snd_register_oss_device(int type, struct snd_card > > *card, int dev, > > int register1 = -1, register2 = -1; > > struct device *carddev = snd_card_get_device_link(card); > > > > - if (card && card->number >= 8) > > + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) > > return 0; /* ignore silently */ > > if (minor < 0) > > return minor; > > @@ -170,7 +170,7 @@ int snd_unregister_oss_device(int type, struct > > snd_card *card, int dev) > > int track2 = -1; > > struct snd_minor *mptr; > > > > - if (card && card->number >= 8) > > + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) > > return 0; > > if (minor < 0) > > return minor; > > -- > > 1.7.7 > > > >

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[alsa-devel] soc-dsp questions
by Vinod Koul 09 Nov '11

09 Nov '11

Hey Liam, I had few questions on your dsp work, would be great if you could answer them and help me in understanding the approach. - Is it possible to have one FE connected to multiple BEs (voice FE to modem and headset bBE), if so how? - SND_SOC_DAI_LINK_OPT_HOST, what does this mean, does not seem to be used anywhere in core? - Why comment need dummy codec for some BEs like BT etc, when soc-core has already one? - any reason why be haves a number associated with a BE, than usual convention of referring things by string name in asoc? and last one, when would this be merged to mainline, 3.3? -- ~Vinod

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[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for Toshiba T110
by David Henningsson 09 Nov '11

09 Nov '11

According to the bug reporter, model=auto is needed to make the internal microphone work. BugLink: https://bugs.launchpad.net/bugs/819699 Reported-by: Andrej (agno01) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e706e4..0de2119 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3062,7 +3062,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), - SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5066_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), -- 1.7.5.4

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[alsa-devel] [RFC] hda - is_jack_detectable()
by Raymond Yau 09 Nov '11

09 Nov '11

Refer to HDA Specification 7.3.3.31 Configuration Default Misc[3:0] is a bit field used to indicate other information about the jack. Currently, only bit 0 is defined. If bit 0 is set, it indicates that the jack has no presence detect capability, so even if a Pin Complex indicates that the codec hardware supports the presence detect functionality on the jack, the external circuitry is not capable of supporting the functionality. The bit definitions for the Miscfield are in Table 106. Do we need to change is_jack_detectable() when user setup BIOS to use AC97 Front Audio Panel which set AC_DEFCFG_MISC_NO_PRESENCE bit ? +#define get_defcfg_misc(cfg) \ + ((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) { return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) && + (!(get_defcfg_misc(codec,nid) & AC_DEFCFG_MISC_NO_PRESENCE)) && (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP); } or add another fuction is_defcfg_misc_no_presence() to indicate jack has no presence detect capability ?

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[alsa-devel] the monitors with working Displayport audio
by Wu Fengguang 09 Nov '11

09 Nov '11

Hi Takashi, I enumerated all the DELL DisplayPort monitors in our lab and find only two of them being able to produce audible DisplayPort audio... - DELL 2408WFP - DELL 2709W - DELL 3008WFP no CEA extension found - DELL U2410 ELD reports OK but cannot hear DP sound monitor_present 1 eld_valid 1 monitor_name DELL U2410 connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xac10 product_id 0xf017 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0x1] FL/FR sad_count 1 sad0_coding_type [0x1] LPCM sad0_channels 2 sad0_rates [0xe0] 32000 44100 48000 sad0_bits [0xe0000] 16 20 24 - DELL U3011 ELD and DP sound both OK monitor_present 1 eld_valid 1 monitor_name DELL U3011 connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xac10 product_id 0x4065 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0xf] FL/FR LFE FC RL/RR sad_count 1 sad0_coding_type [0x1] LPCM sad0_channels 6 sad0_rates [0x1ee0] 32000 44100 48000 96000 176400 192000 384000 sad0_bits [0xe0000] 16 20 24 - Eizo FlexScan EV2333W ELD and DP sound both OK monitor_present 1 eld_valid 1 monitor_name EV2333W connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xc315 product_id 0x2068 port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0x1] FL/FR sad_count 1 sad0_coding_type [0x1] LPCM sad0_channels 2 sad0_rates [0x1ee0] 32000 44100 48000 96000 176400 192000 384000 sad0_bits [0xe0000] 16 20 24 Thanks, Fengguang

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Re: [alsa-devel] Why my linux OS has only 8 sound cards ? How can I modify it ?
by Linus Torvalds 08 Nov '11

08 Nov '11

Chris - you need to pick a better set of people to ask. A mailing list, right maintainer etc. Takashi, any idea? Do we have some 8-sound-card limit somewhere? I do see that we have that odd if (card && card->number >= 8) return 0; /* ignore silently */ in sound/core/sound_oss.c: snd_register_oss_device(), but it has no comments. The thing goes back to 2005 with the commit log entry of "[ALSA] dynamic minors (6/6): increase maximum number of sound cards", but the value '8'isn't explained. Linus On Tue, Nov 8, 2011 at 12:30 AM, Chris Shen <chris2shen(a)gmail.com> wrote: > Hi,everyone! > My version of kernel is linux-2.6.35 .And my board has 10 sounds > cards.After system was up,I used the comand "aplay -l ".Then console > would display this: > **** List of PLAYBACK Hardware Devices **** > card 0: imx3stack [imx-3stack], device 0: WM8731 WM8731-0 [] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: default [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 2: default_1 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 3: default_2 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 4: default_3 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 5: default_4 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 6: default_5 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 7: default_6 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 8: default_7 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 9: default_8 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > But,when I used the command "ls /dev/dsp*" ,the console display: > /dev/dsp /dev/dsp2 /dev/dsp4 /dev/dsp6 > /dev/dsp1 /dev/dsp3 /dev/dsp5 /dev/dsp7 > when the board is up ,I found have this err "unable to register OSS > mixer device 8:0" and "unable to register OSS mixer device 9:0" . > I think the kernel's max sound cards is 8 .How can I modify it ? > Thank you! >

3 3

[alsa-devel] [regression] probably in mixer.c code
by Alexey Fisher 08 Nov '11

08 Nov '11

Hallo all, Takashi, there is one regression in current linux main tree: cval->res quirks do not work any more. See usb/mixer.c:1150 Are there was some work in this area or may be some one bisected this bug down? Regards, Alexey.

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 08 Nov '11

08 Nov '11

The following changes since commit 1ea6b8f48918282bdca0b32a34095504ee65bab5: Linux 3.2-rc1 (2011-11-07 16:16:02 -0800) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.3 Axel Lin (6): ASoC: wm8400: Fix setting Fout clock divider for FLL Control 4 ASoC: wm8996: Avoid a redundant i2c_get_clientdata call in wm8996_i2c_remove ASoC: wm8940: Fix setting PLL Output clock division ratio ASoC: wm8940: Fix a typo for the mask of setting WM8940_BCLKDIV ASoC: tlv320aic3x: Use snd_soc_update_bits for read-modify-write ASoC: Avoid a redundant read in cs42l51_pdn_event Kuninori Morimoto (3): ASoC: fsi: fixup compile warning ASoC: ak4642: ak4642 was tested ASoC: fsi: add valid data position control support Mark Brown (29): ASoC: Convert wm8995 MICBIASes to supply widgets ASoC: Convert wm8991 MICBIAS to a supply widget ASoC: Convert wm8990 MICBIAS to a supply widget ASoC: Convert wm8988 MICBIAS to a supply widget ASoC: Convert wm8985 MICBIAS to a supply widget ASoC: Convert wm8983 MICBIAS to a supply widget ASoC: Convert wm8974 MICBIAS to a supply widget ASoC: Convert wm8961 MICBIAS to a supply widget ASoC: Convert wm8960 MICBIAS to a supply widget ASoC: Convert wm8904 MICBIAS to a supply widget ASoC: Convert wm8900 MICBIAS to a supply widget ASoC: Convert wm8400 MICBIAS to a supply widget ASoC: Start WM8962 FLL if SYSCLK is enabled ASoC: Enable SYSCLK last when enabling WM8962 mic detection ASoC: WM8962 accessory detection requires MICBIAS ASoC: Disable MICBIAS and SYSCLK when stopping WM8962 accessory detection ASoC: Sort LM4857 with the CODECs in the Makefile ASoC: Fix sort of jz4740 in Makefile ASoC: Update git repository URL ASoC: Remove needless unlikely() ASoC: Only enable thermal shutdown when required on WM9081 ASoC: Manage thermal shutdown for WM8962 ASoC: Disable thermal shutdown when not using speakers in wm_hubs ASoC: Don't use wm8994->control_data in wm8994_readable_register() ASoC: Don't use wm8994->control_data when requesting IRQs ASoC: Ensure we get an impedence reported for WM8958 jack detect Merge branch 'for-3.2' into for-3.3 Merge branch 'for-3.2' of git://git.kernel.org/.../broonie/sound into for-3.2 Merge branch 'for-3.2' into for-3.3 ramesh.babu(a)linux.intel.com (1): ASoC: Allow machines to ignore pmdown_time per-link MAINTAINERS | 2 +- include/linux/mfd/wm8994/registers.h | 15 ++++++++++++ include/sound/soc.h | 3 ++ sound/soc/codecs/Makefile | 6 ++-- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/cs42l51.c | 13 ++++------ sound/soc/codecs/tlv320aic3x.c | 41 +++++++++++--------------------- sound/soc/codecs/wm8400.c | 6 ++-- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 4 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 20 +++++++++++++++ sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8983.c | 3 +- sound/soc/codecs/wm8985.c | 3 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm8991.c | 4 +- sound/soc/codecs/wm8994.c | 43 +++++++++++++++++++++++----------- sound/soc/codecs/wm8995.c | 6 +++- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 + sound/soc/codecs/wm_hubs.c | 3 ++ sound/soc/sh/fsi.c | 21 ++++++++++++++-- sound/soc/soc-pcm.c | 3 +- 27 files changed, 140 insertions(+), 78 deletions(-)

2 1

[alsa-devel] ASoC updates for 3.2
by Mark Brown 08 Nov '11

08 Nov '11

The following changes since commit 1ea6b8f48918282bdca0b32a34095504ee65bab5: Linux 3.2-rc1 (2011-11-07 16:16:02 -0800) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 This is mostly a merge up of 3.2 rather than new code. Mark Brown (5): ASoC: Update git repository URL ASoC: Don't use wm8994->control_data in wm8994_readable_register() ASoC: Don't use wm8994->control_data when requesting IRQs ASoC: Ensure we get an impedence reported for WM8958 jack detect Merge branch 'for-3.2' of git://git.kernel.org/.../broonie/sound into for-3.2 MAINTAINERS | 2 +- include/linux/mfd/wm8994/registers.h | 15 ++++++++++++ sound/soc/codecs/wm8994.c | 43 +++++++++++++++++++++++----------- 3 files changed, 45 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 1/1] alsa-driver: RHEL 6.1 compilation fix
by Konstantin Ozerkov 08 Nov '11

08 Nov '11

RHEL 6.1 (2.6.32-132.*) kernels already have hex_to_bin() function. Tune check in the alsa-driver/include/adriver.h Tested on Scientific Linux 6.1 (RHEL 6.1 clone), kernels: 2.6.32-131.0.15.el6.x86_64 (original, from DVD installation) 2.6.32-131.17.1.el6.x86_64 (last security update) Signed-off-by: Konstantin Ozerkov <kozerkov(a)parallels.com> Signed-off-by: Denis V. Lunev <den(a)openvz.org> diff --git a/include/adriver.h b/include/adriver.h index d9c7dee..aeb084e 100644 --- a/include/adriver.h +++ b/include/adriver.h @@ -1920,6 +1920,8 @@ blocking_notifier_chain_unregister(struct blocking_notifier_head *nh, /* hex_to_bin() */ #if LINUX_VERSION_CODE < KERNEL_VERSION(2, 6, 35) +/* RHEL 6.1 kernels has version 2.6.32, but already have hex_to_bin() */ +#if !defined(RHEL_RELEASE_CODE) || RHEL_RELEASE_CODE < RHEL_RELEASE_VERSION(6,1) static inline int hex_to_bin(char c) { if (c >= '0' && c <= '9') @@ -1931,6 +1933,7 @@ static inline int hex_to_bin(char c) return -1; } #endif +#endif #ifndef CONFIG_HAVE_VZALLOC #include <linux/vmalloc.h> -- 1.7.1

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[alsa-devel] [PATCH 1/2] ASoC: wm9081: Fix reading wrong register for setting VMID 2*240k
by Axel Lin 08 Nov '11

08 Nov '11

VMID Divider Enable and Select is controlled by BIT[2:1] of WM9081_VMID_CONTROL register (04h). Current code reads wrong register (WM9081_BIAS_CONTROL_1) for setting VMID 2*240k. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9081.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 3cd35a0..fe65618 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -818,7 +818,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_VMID_CONTROL); reg &= ~WM9081_VMID_SEL_MASK; reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Disable debounce on some WM8962 interrupts
by Mark Brown 08 Nov '11

08 Nov '11

Allow them to work when the device is unclocked. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index aa310a2..5d1704d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4059,6 +4059,12 @@ static int wm8962_probe(struct snd_soc_codec *codec) /* Stereo control for EQ */ snd_soc_update_bits(codec, WM8962_EQ1, WM8962_EQ_SHARED_COEFF, 0); + /* Don't debouce interrupts so we don't need SYSCLK */ + snd_soc_update_bits(codec, WM8962_IRQ_DEBOUNCE, + WM8962_FLL_LOCK_DB | WM8962_PLL3_LOCK_DB | + WM8962_PLL2_LOCK_DB | WM8962_TEMP_SHUT_DB, + 0); + wm8962_add_widgets(codec); /* Save boards having to disable DMIC when not in use */ -- 1.7.7.1

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Mark Brown 08 Nov '11

08 Nov '11

On Tue, Nov 08, 2011 at 06:03:53PM +0900, Tomoya MORINAGA wrote: So, I started looking at this but... > struct snd_ml7213i2s_pcm { > enum snd_soc_control_type control_type; > struct snd_ml7213i2s *ml7213i2s; > spinlock_t lock; > unsigned int irq_pos; > unsigned int buf_pos; > struct snd_pcm_substream *substream; > struct cbdata cbd; /* i2s callback info */ > unsigned int channels; > unsigned int rw; > unsigned int rate; > unsigned int ch; > unsigned int setup_flag; > unsigned int format; > unsigned int bclkfs; > struct mutex i2c_mutex; > }; ...this looks *really* confused, there's things in here which are a mix of DMA controller and CODEC driver things. The CODEC and DMA drivers shouldn't know anything about each other, let alone be referencing the same data structure. > /* > * wm8731 register cache > * We can't read the WM8731 register space when we are > * using 2 wire for device control, so we cache them instead. > * There is no point in caching the reset register > */ > static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { > 0x0097, 0x0097, 0x0079, 0x0079, > 0x000a, 0x0008, 0x009f, 0x000a, > 0x0000, 0x0000 > }; This is is just obviously wrong for this driver. I stopped reading the code at this point.

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[alsa-devel] Sound card lx6464es
by mdrslmr＠t-online.de 08 Nov '11

08 Nov '11

Hi, could anyone please provide me with a basic alsa configuration (.asoundrc) for the caard lx6464es for playback and recording? Best regards Michael

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[alsa-devel] spdif: aplay -L output
by ajay khandelwal 08 Nov '11

08 Nov '11

Hi All, When I do aplay -L on my pc it gives following as mentioned below. Name expoted for spdif is iec958:CARD=ICH6,DEV=0 Is this name coming from configuration file or is exported by driver. # /usr/bin/aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=ICH6 Intel ICH6, Intel ICH6 Default Audio Device front:CARD=ICH6,DEV=0 Intel ICH6, Intel ICH6 Front speakers surround40:CARD=ICH6,DEV=0 Intel ICH6, Intel ICH6 4.0 Surround output to Front and Rear speakers surround41:CARD=ICH6,DEV=0 Intel ICH6, Intel ICH6 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=ICH6,DEV=0 Intel ICH6, Intel ICH6 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=ICH6,DEV=0 Intel ICH6, Intel ICH6 5.1 Surround output to Front, Center, Rear and Subwoofer speakers iec958:CARD=ICH6,DEV=0 Intel ICH6, Intel ICH6 - IEC958 IEC958 (S/PDIF) Digital Audio Output Best Regards, Ajay

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[alsa-devel] SPDIF: How to play mono sound
by ajay khandelwal 08 Nov '11

08 Nov '11

Hi all, I have a pc with SPDIF out. I want to play audio file with single channel(mono) using aplay. How to do that, when I try to play with following command, it gives error aplay -Diec958:CARDICH6,DEV=0 mono_music.wav it gives following error aplay: set_param:1065: Channels count not available. Thanks and Regards, Ajay

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[alsa-devel] Swapped channels issue on pxa-ssp based platforms
by Daniel Mack 08 Nov '11

08 Nov '11

Hi, we're hunting a weird bug on PXA3xx/ssp based audio devices which causes the playback channels to be swapped when the capture stream is started at the same time. Content for the left channel appears on the right and vice versa, in approximately 50% of all test runs of the following command: $ arecord -f cd - -D hw:0,0 | aplay -D hw:0,0 -f cd - When just aplay is used, everything's fine. It's unrelated to the codec part, as we see this with machines that feature different models. I guess this is some sort of a race condition in the stream startup, and suspected sound/soc/pxa/pxa-ssp.c to lack some locking, so I added a spinlock around all register read-modify-write cycles. But that doesn't seem to be the reason. It could also be the DMA engine that writes to the wrong position, but I fail to see it right now. Does anyone have an idea where to look for the culprit? Have there been similar effects in other SoC platforms? Thanks for sharing your thoughts :) Daniel

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[alsa-devel] [PATCH] ASoC: Avoid a redundant read in cs42l51_pdn_event
by Axel Lin 08 Nov '11

08 Nov '11

snd_soc_update_bits already does read-modify-write, no need to read the register before calling snd_soc_update_bits. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/cs42l51.c | 13 +++++-------- 1 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 8c3c820..00718b5 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -175,21 +175,18 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - unsigned long value; - - value = snd_soc_read(w->codec, CS42L51_POWER_CTL1); - value &= ~CS42L51_POWER_CTL1_PDN; - switch (event) { case SND_SOC_DAPM_PRE_PMD: - value |= CS42L51_POWER_CTL1_PDN; + snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + CS42L51_POWER_CTL1_PDN, + CS42L51_POWER_CTL1_PDN); break; default: case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + CS42L51_POWER_CTL1_PDN, 0); break; } - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, - CS42L51_POWER_CTL1_PDN, value); return 0; } -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: fsi: fixup compile warning
by Kuninori Morimoto 08 Nov '11

08 Nov '11

This patch fixup below warning ${linux}/sound/soc/sh/fsi.c:442:3:\ warning: passing argument 1 of '__fsi_reg_read' makes pointer\ from integer without a cast ${linux}/sound/soc/sh/fsi.c:517:3: \ warning: passing argument 1 of '__fsi_reg_write' makes pointer\ from integer without a cast ${linux}/sound/soc/sh/fsi.c:663:3: \ warning: passing argument 1 of '__fsi_reg_mask_set' makes pointer\ from integer without a cast Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a32fd16..809f083 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -234,13 +234,13 @@ static void __fsi_reg_mask_set(u32 __iomem *reg, u32 mask, u32 data) } #define fsi_reg_write(p, r, d)\ - __fsi_reg_write((u32)(p->base + REG_##r), d) + __fsi_reg_write((p->base + REG_##r), d) #define fsi_reg_read(p, r)\ - __fsi_reg_read((u32)(p->base + REG_##r)) + __fsi_reg_read((p->base + REG_##r)) #define fsi_reg_mask_set(p, r, m, d)\ - __fsi_reg_mask_set((u32)(p->base + REG_##r), m, d) + __fsi_reg_mask_set((p->base + REG_##r), m, d) #define fsi_master_read(p, r) _fsi_master_read(p, MST_##r) #define fsi_core_read(p, r) _fsi_master_read(p, p->core->r) -- 1.7.5.4

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Mark Brown 08 Nov '11

08 Nov '11

On Tue, Nov 08, 2011 at 06:03:53PM +0900, Tomoya MORINAGA wrote: > Though roughly, we re-created 3 files codec driver, platform drive > and machine driver. (Not debugged/tested) > Could you check these files ? Please send patches for review using the standard kernel review process - see Documentation/SubmittingPatches. Sending a single enormous review makes it much more difficult for people to review as there's so much code in the one mail and attachments don't work with people's normal workflows.

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Re: [alsa-devel] [PATCH 1/1] ALSA: intel8x0: Improve comments for VM optimization
by Takashi Iwai 08 Nov '11

08 Nov '11

At Mon, 7 Nov 2011 20:33:25 +0400, Denis V. Lunev wrote: > > The recently merged 228cf79376f1 looks a bit hackish while it is not. > The change was quite simple. In a virtualized environment the > patch unhacks old kludge introduced for old broken AC97 hardware. > > This patch adds proper comment to "unkludge" code. > > Signed-off-by: Denis V. Lunev <den(a)openvz.org> > Signed-off-by: Konstantin Ozerkov <kozerkov(a)parallels.com> Applied now. Thanks. Takashi > --- > sound/pci/intel8x0.c | 7 +++++++ > 1 files changed, 7 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c > index 29e3125..c3b9bd0 100644 > --- a/sound/pci/intel8x0.c > +++ b/sound/pci/intel8x0.c > @@ -1077,6 +1077,13 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs > } > if (civ != igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV)) > continue; > + > + /* IO read operation is very expensive inside virtual machine > + * as it is emulated. The probability that subsequent PICB read > + * will return different result is high enough to loop till > + * timeout here. > + * Same CIV is strict enough condition to be sure that PICB > + * is valid inside VM on emulated card. */ > if (chip->inside_vm) > break; > if (ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb)) > -- > 1.7.4.4 >

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[alsa-devel] ALSA Snapshot alsa-utils-1.0.24.2.25.g9fa7c No user enum control support in alsa-lib
by b0h3m4 07 Nov '11

07 Nov '11

Hi there I'm trying to compile snapshot of ALSA packages on Oneiric x86 for via8235 >alsa-driver-1.0.24.72.g7c48f.1314.geb1d0.tar.bz2 >alsa-lib-1.0.24.1.46.g15c6f.tar.bz2 >alsa-utils-1.0.24.2.25.g9fa7c.tar.bz2 First two packages go without a hitch on alsa-utils I'm stuck at >checking for snd_ctl_elem_add_enumerated... no >configure: error: No user enum control support in alsa-lib Anything I should be aware of?

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Re: [alsa-devel] Wierd hack to sound/pci/intel8x0.c
by Takashi Iwai 07 Nov '11

07 Nov '11

At Mon, 07 Nov 2011 11:35:49 +0100, Gerd Hoffmann wrote: > > Hi, > > > We discuss Intel ICH/AC'97 (snd-intel8x0) here, but I hope that PCI SSID > > is same. > > Hmm, it's not, the ac97 emulation doesn't use the default qemu subsystem > id for some reason. Here is the entry: > > [root@fedora ~]# lspci -vns6 > 00:06.0 0401: 8086:2415 (rev 01) > Subsystem: 8086:0000 > Physical Slot: 6 > Flags: bus master, medium devsel, latency 0, IRQ 10 > I/O ports at c000 [size=1K] > I/O ports at c400 [size=256] > Kernel driver in use: Intel ICH > Kernel modules: snd-intel8x0 > > I guess that should be fixed. 8086:0000 isn't valid anyway and I doubt > it serves any useful purpose other than avoiding a guest complaining > about the subsystem id being unset. Yeah, especially the value 0 looks pretty bad. > /me goes preparing a patch. Thanks! Takashi

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Re: [alsa-devel] Wierd hack to sound/pci/intel8x0.c
by Takashi Iwai 07 Nov '11

07 Nov '11

At Mon, 07 Nov 2011 10:25:24 +0100, Gerd Hoffmann wrote: > > Hi, > > > Agreed. If we can know the virtual device for KVM in a better way > > (e.g. any specific PCI SSID or such), we can narrow the condition more > > safely. > > PCI Subsystem ID 1af4:1100 is qemu/kvm (not only Intel HDA, most other > emulated pci devices have it too). Complete entry: > > 00:04.0 0403: 8086:2668 (rev 01) > Subsystem: 1af4:1100 > Physical Slot: 4 > Flags: bus master, fast devsel, latency 0, IRQ 11 > Memory at e2030000 (32-bit, non-prefetchable) [size=16K] > Kernel driver in use: HDA Intel > Kernel modules: snd-hda-intel Thanks. BTW, this reminds me of a possible improvement in the current HD-audio driver. It has some workaround for the delayed DMA-position update (the DMA-position isn't completely updated when IRQ is issued) by offsetting the buffer position. This is controlled by bdl_pos_adj option of snd-hda-intel driver, and this could be zero for VMs. Takashi

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Re: [alsa-devel] Wierd hack to sound/pci/intel8x0.c
by Takashi Iwai 06 Nov '11

06 Nov '11

At Sun, 06 Nov 2011 18:31:42 +0200, Avi Kivity wrote: > > On 11/06/2011 06:15 PM, Denis V. Lunev wrote: > > On 11/6/11 6:51 PM, Avi Kivity wrote: > >> The recently merged 228cf79376f1 ("ALSA: intel8x0: Improve performance > >> in virtual environment") is hacky and somewhat wrong. > >> > >> First, the detection code > >> > >> + if (inside_vm< 0) { > >> + /* detect KVM and Parallels virtual environments */ > >> + inside_vm = kvm_para_available(); > >> +#if defined(__i386__) || defined(__x86_64__) > >> + inside_vm = inside_vm || > >> boot_cpu_has(X86_FEATURE_HYPERVISOR); > >> +#endif > >> + } > >> + > >> > >> is incorrect. It detects that you're running in a guest, but that > >> doesn't imply that the device you're accessing is emulated. It may be a > >> host device assigned to the guest; presumably the optimization you apply > >> doesn't work for real devices. > >> > >> Second, the optimization itself looks fishy: > >> > >> spin_lock(&chip->reg_lock); > >> do { > >> civ = igetbyte(chip, ichdev->reg_offset + > >> ICH_REG_OFF_CIV); > >> ptr1 = igetword(chip, ichdev->reg_offset + > >> ichdev->roff_picb); > >> position = ichdev->position; > >> if (ptr1 == 0) { > >> udelay(10); > >> continue; > >> } > >> - if (civ == igetbyte(chip, ichdev->reg_offset + > >> ICH_REG_OFF_CIV)&& > >> - ptr1 == igetword(chip, ichdev->reg_offset + > >> ichdev->roff_picb)) > >> + if (civ != igetbyte(chip, ichdev->reg_offset + > >> ICH_REG_OFF_CIV)) > >> + continue; > >> + if (chip->inside_vm) > >> + break; > >> + if (ptr1 == igetword(chip, ichdev->reg_offset + > >> ichdev->roff_picb)) > >> break; > >> } while (timeout--); > >> > >> > >> Why is the emulated device timing out? Can't the emulation be fixed to > >> behave like real hardware? > >> > >> Last, please copy kvm(a)vger.kernel.org on such issues. > >> > > The problem is that emulation can not be fixed. > > > > How this is working for real hardware? You get data from real sound > > card register. > > The scheduling is off at the moment thus you can not be re-scheduled. > > > > In the virtual environment the situation is different. Any IO > > emulation is expensive. > > The processor is switched from guest to hypervisor and further to > > emulation process > > takes a lot of time. This time is enough to obtain different value on > > next register read. > > That's why this code is really timed out. Please also note that host > > scheduler also > > plays his games and could schedule out VCPU thread. > > > > Note on kvm this is rare, since the guest thread and the emulator thread > are the same. > > > The problem could be potentially fixed reducing precision of PICB > > emulation, > > but this results in lower sound quality. > > > > This kludge has been written this way in order not to break legacy > > card for which we > > do not have an access. The code reading PICB/CIV registers second time > > was added > > to fix issues on unknown for now platform and it looks not possible > > how to find/test > > against this platform now. We have checked Windows drivers written by > > Intel/AMD > > (32/64 bit) and MacOS ones. There is no second reading of CIV/PICB > > inside. We > > hope that this is relay needed only for some rare hadware devices. > > > > Ok, so if I understand correctly, this loop is a hack for broken > hardware, and this patch basically unhacks it back, assuming that the > emulated (or assigned) hardware is not broken. Exactly. The loop itself is still needed, but the check can be less restrictive. > > The only thing we can is to improve detection code. Suggestions are > > welcome. > > I think it's fine to assume that you're not assigning a 2004 era sound > card to a guest. So I think the code is fine as it is, and can only > suggest to add a comment explaining the mess. True. Denis, care to submit a patch? thanks, Takashi

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Re: [alsa-devel] Wierd hack to sound/pci/intel8x0.c
by Takashi Iwai 06 Nov '11

06 Nov '11

At Sun, 6 Nov 2011 20:15:01 +0400, Denis V. Lunev wrote: > > On 11/6/11 6:51 PM, Avi Kivity wrote: > > The recently merged 228cf79376f1 ("ALSA: intel8x0: Improve performance > > in virtual environment") is hacky and somewhat wrong. > > > > First, the detection code > > > > + if (inside_vm< 0) { > > + /* detect KVM and Parallels virtual environments */ > > + inside_vm = kvm_para_available(); > > +#if defined(__i386__) || defined(__x86_64__) > > + inside_vm = inside_vm || > > boot_cpu_has(X86_FEATURE_HYPERVISOR); > > +#endif > > + } > > + > > > > is incorrect. It detects that you're running in a guest, but that > > doesn't imply that the device you're accessing is emulated. It may be a > > host device assigned to the guest; presumably the optimization you apply > > doesn't work for real devices. > > > > Second, the optimization itself looks fishy: > > > > spin_lock(&chip->reg_lock); > > do { > > civ = igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV); > > ptr1 = igetword(chip, ichdev->reg_offset + > > ichdev->roff_picb); > > position = ichdev->position; > > if (ptr1 == 0) { > > udelay(10); > > continue; > > } > > - if (civ == igetbyte(chip, ichdev->reg_offset + > > ICH_REG_OFF_CIV)&& > > - ptr1 == igetword(chip, ichdev->reg_offset + > > ichdev->roff_picb)) > > + if (civ != igetbyte(chip, ichdev->reg_offset + > > ICH_REG_OFF_CIV)) > > + continue; > > + if (chip->inside_vm) > > + break; > > + if (ptr1 == igetword(chip, ichdev->reg_offset + > > ichdev->roff_picb)) > > break; > > } while (timeout--); > > > > > > Why is the emulated device timing out? Can't the emulation be fixed to > > behave like real hardware? > > > > Last, please copy kvm(a)vger.kernel.org on such issues. > > > The problem is that emulation can not be fixed. > > How this is working for real hardware? You get data from real sound card > register. > The scheduling is off at the moment thus you can not be re-scheduled. > > In the virtual environment the situation is different. Any IO emulation > is expensive. > The processor is switched from guest to hypervisor and further to > emulation process > takes a lot of time. This time is enough to obtain different value on > next register read. > That's why this code is really timed out. Please also note that host > scheduler also > plays his games and could schedule out VCPU thread. > > The problem could be potentially fixed reducing precision of PICB emulation, > but this results in lower sound quality. > > This kludge has been written this way in order not to break legacy card > for which we > do not have an access. The code reading PICB/CIV registers second time > was added > to fix issues on unknown for now platform and it looks not possible how > to find/test > against this platform now. We have checked Windows drivers written by > Intel/AMD > (32/64 bit) and MacOS ones. There is no second reading of CIV/PICB > inside. We > hope that this is relay needed only for some rare hadware devices. > > The only thing we can is to improve detection code. Suggestions are welcome. Agreed. If we can know the virtual device for KVM in a better way (e.g. any specific PCI SSID or such), we can narrow the condition more safely. thanks, Takashi

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[alsa-devel] [PATCH] ALSA: control: remove compilation warning on 32-bit
by Olof Johansson 06 Nov '11

06 Nov '11

This was introduced by 'ALSA: control: add support for ENUMERATED user space controls' which adds a u64 variable that gets cast to a pointer: sound/core/control.c: In function 'snd_ctl_elem_init_enum_names': sound/core/control.c:1089: warning: cast to pointer from integer of different size Since we don't support 64-bit userspace on 32-bit kernel, it's safe to cast to long before casting to pointer to avoid the warning. Signed-off-by: Olof Johansson <olof(a)lixom.net> --- sound/core/control.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 978fe1a..d09c335 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1081,12 +1081,12 @@ static int snd_ctl_elem_init_enum_names(struct user_element *ue) char *names, *p; size_t buf_len, name_len; unsigned int i; + const long user_ptrval = ue->info.value.enumerated.names_ptr; if (ue->info.value.enumerated.names_length > 64 * 1024) return -EINVAL; - names = memdup_user( - (const void __user *)ue->info.value.enumerated.names_ptr, + names = memdup_user((const void __user *)user_ptrval, ue->info.value.enumerated.names_length); if (IS_ERR(names)) return PTR_ERR(names); -- 1.7.4.1

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[alsa-devel] [PATCH v2 RESEND] ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
by Axel Lin 05 Nov '11

05 Nov '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- RESEND: I just found I doesn't set my mail client properly to show correct sender, so here is a resend. v2: For Initial cold start ( change bias level from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY ), the v1 does not set VMID_SEL[1:0] bits correctly for Normal bias enable & soft start off. In v2 we also clear VMID_SEL[1:0] bits for Normal bias enable & soft start off. We did clear VMID_SEL[1:0] bits in SND_SOC_BIAS_OFF, thus we know the VMID_SEL[1:0] bits status is 0b00 in this case. sound/soc/codecs/wm9081.c | 82 +++++++++++++++++++------------------------- 1 files changed, 35 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 66092ef..9c38ddb 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -766,84 +766,73 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); - + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | 0x6, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg |= WM9081_BIAS_ENA; - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_BIAS_ENA | + WM9081_VMID_SEL_MASK, + WM9081_BIAS_ENA); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, WM9081_BIAS_SRC); /* Disable VMID and biases with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK | + WM9081_BIAS_ENA, WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1242,11 +1231,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); -- 1.7.5.4

3 3

[alsa-devel] [PATCH] ASoC: Disable thermal shutdown when not using speakers in wm_hubs
by Mark Brown 05 Nov '11

05 Nov '11

The thermal shutdown support in wm_hubs devices is tied to the speaker drivers (which are the only high power subsystems within the device). Ensure minimal current usage when the thermal shutdown support is not required by disabling the circuit when the speaker drivers are powered down. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4..f98170e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -654,6 +654,7 @@ SND_SOC_DAPM_MIXER("SPKL Boost", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0, right_speaker_boost, ARRAY_SIZE(right_speaker_boost)), +SND_SOC_DAPM_SUPPLY("TSHUT", WM8993_POWER_MANAGEMENT_2, 14, 0, NULL, 0), SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, NULL, 0), SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, @@ -789,10 +790,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKL Driver", NULL, "VMID" }, { "SPKL Driver", NULL, "SPKL Boost" }, { "SPKL Driver", NULL, "CLK_SYS" }, + { "SPKL Driver", NULL, "TSHUT" }, { "SPKR Driver", NULL, "VMID" }, { "SPKR Driver", NULL, "SPKR Boost" }, { "SPKR Driver", NULL, "CLK_SYS" }, + { "SPKR Driver", NULL, "TSHUT" }, { "SPKOUTLP", NULL, "SPKL Driver" }, { "SPKOUTLN", NULL, "SPKL Driver" }, -- 1.7.7.1

1 0

[alsa-devel] [PATCH] ASoC: Only enable thermal shutdown when required on WM9081
by Mark Brown 05 Nov '11

05 Nov '11

The WM9081 thermal shutdown is only effective when the speaker output is enabled so disable it when that is not in use for a small current saving. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 550bf92..a4570e7 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -796,6 +796,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0), }; @@ -818,6 +819,7 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "Speaker PGA", NULL, "CLK_SYS" }, { "Speaker", NULL, "Speaker PGA" }, + { "Speaker", NULL, "TSENSE" }, { "SPKN", NULL, "Speaker" }, { "SPKP", NULL, "Speaker" }, -- 1.7.7.1

1 0

[alsa-devel] [PATCH] ASoC: Convert WM9081 to direct regmap API usage
by Mark Brown 04 Nov '11

04 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 280 +++++++++++++++++++++++++++++---------------- 1 files changed, 182 insertions(+), 98 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 4e0489f..09d6da0 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -19,6 +19,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -30,69 +31,63 @@ #include <sound/wm9081.h> #include "wm9081.h" -static u16 wm9081_reg_defaults[] = { - 0x0000, /* R0 - Software Reset */ - 0x0000, /* R1 */ - 0x00B9, /* R2 - Analogue Lineout */ - 0x00B9, /* R3 - Analogue Speaker PGA */ - 0x0001, /* R4 - VMID Control */ - 0x0068, /* R5 - Bias Control 1 */ - 0x0000, /* R6 */ - 0x0000, /* R7 - Analogue Mixer */ - 0x0000, /* R8 - Anti Pop Control */ - 0x01DB, /* R9 - Analogue Speaker 1 */ - 0x0018, /* R10 - Analogue Speaker 2 */ - 0x0180, /* R11 - Power Management */ - 0x0000, /* R12 - Clock Control 1 */ - 0x0038, /* R13 - Clock Control 2 */ - 0x4000, /* R14 - Clock Control 3 */ - 0x0000, /* R15 */ - 0x0000, /* R16 - FLL Control 1 */ - 0x0200, /* R17 - FLL Control 2 */ - 0x0000, /* R18 - FLL Control 3 */ - 0x0204, /* R19 - FLL Control 4 */ - 0x0000, /* R20 - FLL Control 5 */ - 0x0000, /* R21 */ - 0x0000, /* R22 - Audio Interface 1 */ - 0x0002, /* R23 - Audio Interface 2 */ - 0x0008, /* R24 - Audio Interface 3 */ - 0x0022, /* R25 - Audio Interface 4 */ - 0x0000, /* R26 - Interrupt Status */ - 0x0006, /* R27 - Interrupt Status Mask */ - 0x0000, /* R28 - Interrupt Polarity */ - 0x0000, /* R29 - Interrupt Control */ - 0x00C0, /* R30 - DAC Digital 1 */ - 0x0008, /* R31 - DAC Digital 2 */ - 0x09AF, /* R32 - DRC 1 */ - 0x4201, /* R33 - DRC 2 */ - 0x0000, /* R34 - DRC 3 */ - 0x0000, /* R35 - DRC 4 */ - 0x0000, /* R36 */ - 0x0000, /* R37 */ - 0x0000, /* R38 - Write Sequencer 1 */ - 0x0000, /* R39 - Write Sequencer 2 */ - 0x0002, /* R40 - MW Slave 1 */ - 0x0000, /* R41 */ - 0x0000, /* R42 - EQ 1 */ - 0x0000, /* R43 - EQ 2 */ - 0x0FCA, /* R44 - EQ 3 */ - 0x0400, /* R45 - EQ 4 */ - 0x00B8, /* R46 - EQ 5 */ - 0x1EB5, /* R47 - EQ 6 */ - 0xF145, /* R48 - EQ 7 */ - 0x0B75, /* R49 - EQ 8 */ - 0x01C5, /* R50 - EQ 9 */ - 0x169E, /* R51 - EQ 10 */ - 0xF829, /* R52 - EQ 11 */ - 0x07AD, /* R53 - EQ 12 */ - 0x1103, /* R54 - EQ 13 */ - 0x1C58, /* R55 - EQ 14 */ - 0xF373, /* R56 - EQ 15 */ - 0x0A54, /* R57 - EQ 16 */ - 0x0558, /* R58 - EQ 17 */ - 0x0564, /* R59 - EQ 18 */ - 0x0559, /* R60 - EQ 19 */ - 0x4000, /* R61 - EQ 20 */ +static struct reg_default wm9081_reg[] = { + { 0, 0x9081 }, /* R0 - Software Reset */ + { 1, 0x0000 }, /* R1 */ + { 2, 0x00B9 }, /* R2 - Analogue Lineout */ + { 3, 0x00B9 }, /* R3 - Analogue Speaker PGA */ + { 4, 0x0001 }, /* R4 - VMID Control */ + { 5, 0x0068 }, /* R5 - Bias Control 1 */ + { 6, 0x0000 }, /* R6 */ + { 7, 0x0000 }, /* R7 - Analogue Mixer */ + { 8, 0x0000 }, /* R8 - Anti Pop Control */ + { 9, 0x01DB }, /* R9 - Analogue Speaker 1 */ + { 10, 0x0018 }, /* R10 - Analogue Speaker 2 */ + { 11, 0x0180 }, /* R11 - Power Management */ + { 12, 0x0000 }, /* R12 - Clock Control 1 */ + { 13, 0x0038 }, /* R13 - Clock Control 2 */ + { 14, 0x4000 }, /* R14 - Clock Control 3 */ + { 16, 0x0000 }, /* R16 - FLL Control 1 */ + { 17, 0x0200 }, /* R17 - FLL Control 2 */ + { 18, 0x0000 }, /* R18 - FLL Control 3 */ + { 19, 0x0204 }, /* R19 - FLL Control 4 */ + { 20, 0x0000 }, /* R20 - FLL Control 5 */ + { 22, 0x0000 }, /* R22 - Audio Interface 1 */ + { 23, 0x0002 }, /* R23 - Audio Interface 2 */ + { 24, 0x0008 }, /* R24 - Audio Interface 3 */ + { 25, 0x0022 }, /* R25 - Audio Interface 4 */ + { 27, 0x0006 }, /* R27 - Interrupt Status Mask */ + { 28, 0x0000 }, /* R28 - Interrupt Polarity */ + { 29, 0x0000 }, /* R29 - Interrupt Control */ + { 30, 0x00C0 }, /* R30 - DAC Digital 1 */ + { 31, 0x0008 }, /* R31 - DAC Digital 2 */ + { 32, 0x09AF }, /* R32 - DRC 1 */ + { 33, 0x4201 }, /* R33 - DRC 2 */ + { 34, 0x0000 }, /* R34 - DRC 3 */ + { 35, 0x0000 }, /* R35 - DRC 4 */ + { 38, 0x0000 }, /* R38 - Write Sequencer 1 */ + { 39, 0x0000 }, /* R39 - Write Sequencer 2 */ + { 40, 0x0002 }, /* R40 - MW Slave 1 */ + { 42, 0x0000 }, /* R42 - EQ 1 */ + { 43, 0x0000 }, /* R43 - EQ 2 */ + { 44, 0x0FCA }, /* R44 - EQ 3 */ + { 45, 0x0400 }, /* R45 - EQ 4 */ + { 46, 0x00B8 }, /* R46 - EQ 5 */ + { 47, 0x1EB5 }, /* R47 - EQ 6 */ + { 48, 0xF145 }, /* R48 - EQ 7 */ + { 49, 0x0B75 }, /* R49 - EQ 8 */ + { 50, 0x01C5 }, /* R50 - EQ 9 */ + { 51, 0x169E }, /* R51 - EQ 10 */ + { 52, 0xF829 }, /* R52 - EQ 11 */ + { 53, 0x07AD }, /* R53 - EQ 12 */ + { 54, 0x1103 }, /* R54 - EQ 13 */ + { 55, 0x1C58 }, /* R55 - EQ 14 */ + { 56, 0xF373 }, /* R56 - EQ 15 */ + { 57, 0x0A54 }, /* R57 - EQ 16 */ + { 58, 0x0558 }, /* R58 - EQ 17 */ + { 59, 0x0564 }, /* R59 - EQ 18 */ + { 60, 0x0559 }, /* R60 - EQ 19 */ + { 61, 0x4000 }, /* R61 - EQ 20 */ }; static struct { @@ -156,7 +151,7 @@ static struct { }; struct wm9081_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk_source; int mclk_rate; int sysclk_rate; @@ -169,20 +164,84 @@ struct wm9081_priv { struct wm9081_pdata pdata; }; -static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm9081_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM9081_SOFTWARE_RESET: case WM9081_INTERRUPT_STATUS: - return 1; + return true; default: - return 0; + return false; } } -static int wm9081_reset(struct snd_soc_codec *codec) +static bool wm9081_readable_register(struct device *dev, unsigned int reg) { - return snd_soc_write(codec, WM9081_SOFTWARE_RESET, 0); + switch (reg) { + case WM9081_SOFTWARE_RESET: + case WM9081_ANALOGUE_LINEOUT: + case WM9081_ANALOGUE_SPEAKER_PGA: + case WM9081_VMID_CONTROL: + case WM9081_BIAS_CONTROL_1: + case WM9081_ANALOGUE_MIXER: + case WM9081_ANTI_POP_CONTROL: + case WM9081_ANALOGUE_SPEAKER_1: + case WM9081_ANALOGUE_SPEAKER_2: + case WM9081_POWER_MANAGEMENT: + case WM9081_CLOCK_CONTROL_1: + case WM9081_CLOCK_CONTROL_2: + case WM9081_CLOCK_CONTROL_3: + case WM9081_FLL_CONTROL_1: + case WM9081_FLL_CONTROL_2: + case WM9081_FLL_CONTROL_3: + case WM9081_FLL_CONTROL_4: + case WM9081_FLL_CONTROL_5: + case WM9081_AUDIO_INTERFACE_1: + case WM9081_AUDIO_INTERFACE_2: + case WM9081_AUDIO_INTERFACE_3: + case WM9081_AUDIO_INTERFACE_4: + case WM9081_INTERRUPT_STATUS: + case WM9081_INTERRUPT_STATUS_MASK: + case WM9081_INTERRUPT_POLARITY: + case WM9081_INTERRUPT_CONTROL: + case WM9081_DAC_DIGITAL_1: + case WM9081_DAC_DIGITAL_2: + case WM9081_DRC_1: + case WM9081_DRC_2: + case WM9081_DRC_3: + case WM9081_DRC_4: + case WM9081_WRITE_SEQUENCER_1: + case WM9081_WRITE_SEQUENCER_2: + case WM9081_MW_SLAVE_1: + case WM9081_EQ_1: + case WM9081_EQ_2: + case WM9081_EQ_3: + case WM9081_EQ_4: + case WM9081_EQ_5: + case WM9081_EQ_6: + case WM9081_EQ_7: + case WM9081_EQ_8: + case WM9081_EQ_9: + case WM9081_EQ_10: + case WM9081_EQ_11: + case WM9081_EQ_12: + case WM9081_EQ_13: + case WM9081_EQ_14: + case WM9081_EQ_15: + case WM9081_EQ_16: + case WM9081_EQ_17: + case WM9081_EQ_18: + case WM9081_EQ_19: + case WM9081_EQ_20: + return true; + default: + return false; + } +} + +static int wm9081_reset(struct regmap *map) +{ + return regmap_write(map, WM9081_SOFTWARE_RESET, 0x9081); } static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0); @@ -1202,25 +1261,14 @@ static int wm9081_probe(struct snd_soc_codec *codec) int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm9081->control_type); + codec->control_data = wm9081->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET); - if (reg != 0x9081) { - dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg); - ret = -EINVAL; - return ret; - } - - ret = wm9081_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - reg = 0; if (wm9081->pdata.irq_high) reg |= WM9081_IRQ_POL; @@ -1263,15 +1311,9 @@ static int wm9081_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm9081_resume(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; - int i; - - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (i == WM9081_SOFTWARE_RESET) - continue; + struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - snd_soc_write(codec, i, reg_cache[i]); - } + regcache_sync(wm9081->regmap); wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1291,11 +1333,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .set_sysclk = wm9081_set_sysclk, .set_bias_level = wm9081_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm9081_reg_defaults, - .volatile_register = wm9081_volatile_register, - .controls = wm9081_snd_controls, .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, @@ -1304,11 +1341,24 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths), }; +static const struct regmap_config wm9081_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM9081_MAX_REGISTER, + .reg_defaults = wm9081_reg, + .num_reg_defaults = ARRAY_SIZE(wm9081_reg), + .volatile_reg = wm9081_volatile_register, + .readable_reg = wm9081_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm9081_priv *wm9081; + unsigned int reg; int ret; wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL); @@ -1316,7 +1366,30 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm9081); - wm9081->control_type = SND_SOC_I2C; + + wm9081->regmap = regmap_init_i2c(i2c, &wm9081_regmap); + if (IS_ERR(wm9081->regmap)) { + ret = PTR_ERR(wm9081->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + ret = regmap_read(wm9081->regmap, WM9081_SOFTWARE_RESET, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); + goto err_regmap; + } + if (reg != 0x9081) { + dev_err(&i2c->dev, "Device is not a WM9081: ID=0x%x\n", reg); + ret = -EINVAL; + goto err_regmap; + } + + ret = wm9081_reset(wm9081->regmap); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + goto err_regmap; + } if (dev_get_platdata(&i2c->dev)) memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), @@ -1325,13 +1398,24 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) - kfree(wm9081); + goto err_regmap; + + return 0; + +err_regmap: + regmap_exit(wm9081->regmap); +err: + kfree(wm9081); + return ret; } static __devexit int wm9081_i2c_remove(struct i2c_client *client) { + struct wm9081_priv *wm9081 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(wm9081->regmap); kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.1

1 0

[alsa-devel] [PATCH] ASoC: Manage thermal shutdown for WM8962
by Mark Brown 04 Nov '11

04 Nov '11

Disable the thermal shutdown circuits for headphone and speaker when the relevant outputs are not enabled in order to save current in idle modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5202eff..dc85b85 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2692,6 +2692,8 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("TEMP_HP", WM8962_ADDITIONAL_CONTROL_4, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TEMP_SPK", WM8962_ADDITIONAL_CONTROL_4, 1, 0, NULL, 0), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2847,6 +2849,9 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "HPOUTL", NULL, "HPOUT" }, { "HPOUTR", NULL, "HPOUT" }, + + { "HPOUTL", NULL, "TEMP_HP" }, + { "HPOUTR", NULL, "TEMP_HP" }, }; static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { @@ -2863,6 +2868,7 @@ static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { { "Speaker Output", NULL, "Speaker PGA" }, { "Speaker Output", NULL, "SYSCLK" }, { "Speaker Output", NULL, "TOCLK" }, + { "Speaker Output", NULL, "TEMP_SPK" }, { "SPKOUT", NULL, "Speaker Output" }, }; @@ -2891,10 +2897,12 @@ static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { { "SPKOUTL Output", NULL, "SPKOUTL PGA" }, { "SPKOUTL Output", NULL, "SYSCLK" }, { "SPKOUTL Output", NULL, "TOCLK" }, + { "SPKOUTL Output", NULL, "TEMP_SPK" }, { "SPKOUTR Output", NULL, "SPKOUTR PGA" }, { "SPKOUTR Output", NULL, "SYSCLK" }, { "SPKOUTR Output", NULL, "TOCLK" }, + { "SPKOUTR Output", NULL, "TEMP_SPK" }, { "SPKOUTL", NULL, "SPKOUTL Output" }, { "SPKOUTR", NULL, "SPKOUTR Output" }, -- 1.7.7.1

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[alsa-devel] Notes from the Audio BoFs at Prague
by Arun Raghavan 04 Nov '11

04 Nov '11

Hello, We had some pretty productive discussions at the Audio BoFs last week in Prague (since a bunch of us were there for various conferences). I've put up notes from these discussions for those who are interested: http://arunraghavan.net/2011/11/notes-from-the-prague-audio-bofs/ Cheers, Arun p.s.: I've cross-posted to 3 lists, so please be considerate while replying and prune the ones that are not relevant if there is a specific discussion.

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[alsa-devel] need some workarround or explanation.
by Anders Gnistrup 04 Nov '11

04 Nov '11

Hello I have a USB headset, where when plugged in, a volume setting is set on playback and capture channels using the ctl interface for the device. After the volume is set I open the capture interface. but I have i have a 2-3 sec. delay that i just don't understand. I might need to use another solution, but these are difficult to find. The Headset is a plantronics. The code is listed below. #include <alsa/asoundlib.h> #define qDebug printf void set_default_volume(const char *a_hw_dev, int a_percent) { int err; snd_mixer_t *handle; snd_mixer_selem_id_t *sid; snd_mixer_elem_t *elem; snd_mixer_selem_id_alloca(&sid); if ((err = snd_mixer_open(&handle, 0)) < 0) { qDebug("Mixer open error: %s\n", snd_strerror(err)); return; } if ((err = snd_mixer_attach(handle, a_hw_dev)) < 0) { qDebug("Mixer attach %s error: %s\n", a_hw_dev, snd_strerror(err)); snd_mixer_close(handle); return; } if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) { qDebug("Mixer register error: %s\n", snd_strerror(err)); snd_mixer_close(handle); return; } err = snd_mixer_load(handle); if (err < 0) { qDebug("Mixer %s load error: %s\n", a_hw_dev, snd_strerror(err)); snd_mixer_close(handle); return; } for (elem = snd_mixer_first_elem(handle); elem; elem = snd_mixer_elem_next(elem)) { snd_mixer_selem_get_id(elem, sid); qDebug("Simple mixer control '%s',%i\n", snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid)); if (snd_mixer_selem_has_playback_switch(elem)) { snd_mixer_selem_set_playback_switch_all(elem, 1); //unmute } if (snd_mixer_selem_has_playback_volume(elem)) { long min, max; snd_mixer_selem_get_playback_volume_range(elem, &min, &max); qDebug(" pvolume %ld/%ld\n", min , max); snd_mixer_selem_set_playback_volume_all(elem, (a_percent * (max-min))/100); } if (snd_mixer_selem_has_capture_switch(elem)) { snd_mixer_selem_set_capture_switch_all(elem, 1); // unmute all } if (snd_mixer_selem_has_capture_volume(elem)) { long min, max; snd_mixer_selem_get_capture_volume_range(elem, &min, &max); qDebug(" pvolume %ld/%ld\n", min , max); snd_mixer_selem_set_capture_volume_all(elem, (a_percent * (max-min))/100); //set capture volume for all } } snd_mixer_close(handle); return; } int main(int argc, char *argv[]) { snd_pcm_t *m_pcm_handle; int err,count; // Open connection to the PCM devic if ((err = snd_pcm_open(&m_pcm_handle, "plughw:1,0", SND_PCM_STREAM_CAPTURE, 0)) < 0) { for (count=0; count < 5 && err == -EBUSY; count++) { sleep(1); qDebug("AlsaStream:%d open error: %s (%d)\n ", __LINE__, snd_strerror(err), err); err = snd_pcm_open(&m_pcm_handle, "plughw:1,0", SND_PCM_STREAM_CAPTURE, 0); } if (err < 0) { qDebug("AlsaStream:%d open error: %s (%d)\n ", __LINE__, snd_strerror(err), err); return; } if (err>= 0) { printf("Success\n"); } } set_default_volume("hw:1", 20); snd_pcm_close(m_pcm_handle); return 0; }

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[alsa-devel] Question regarding "Fix backport of WM8994 thermal warning" patch
by Thomas Abraham 04 Nov '11

04 Nov '11

Hi Mark, I have a question regarding "Fix backport of WM8994 thermal warning" patch. I am trying sound on Exynos4 smdk board with the latest 3.1 kernel. In the function "wm8994_codec_probe", the kernel crashes due to null pointer when requesting "Thermal Warning" interrupt (line# 3183 of sound/soc/codecs/wm8994.c file). wm8994->control_data is actually NULL at this point. I have not debugged this much but if there is something obvious I am missing, could you please let me know. Thanks, Thomas.

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[alsa-devel] [PATCH 1/2] ASoC: Don't use wm8994->control_data in wm8994_readable_register()
by Mark Brown 04 Nov '11

04 Nov '11

The field is no longer initialised so this will crash if running on wm8958. Reported-by: Thomas Abraham <thomas.abraham(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6b73efd..3d2c3d4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -56,7 +56,7 @@ static int wm8994_retune_mobile_base[] = { static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->control_data; + struct wm8994 *control = codec->control_data; switch (reg) { case WM8994_GPIO_1: -- 1.7.7.1

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[alsa-devel] aplay - memory leak
by Appalayagari Sreedhar 04 Nov '11

04 Nov '11

Hi, I am working with DM368 based custom board. we have external codec interfaced to McBSP. when we run aplay command i am observing memory leak. the leak is around 12k everytime i do aplay. if i get any underruns the leak is more than 12k the kernel version i am working is "Linux-2.6.32.17-davinci1". The codec driver is in alsa model. Could you please help to resolve this problem. Thanks in advance, Sreedhar.

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Re: [alsa-devel] ALSA firewire kernel branch: snd_dice enhancements
by Rolf Anderegg 04 Nov '11

04 Nov '11

On 02.11.2011, at 21:19, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> To get samplerate switching working via ALSA's driver API I have made >> a few changes, especially within the dice_hw_params callback method. >> Plus, I enhanced the rate constraints for the pcm settings. > > On the DICE, the playback and capture streams must use a common rate, so > I thought I'd avoid handling this in the driver and just use whatever > was configured. Isn't common rate for playback & capture stream often required? This could be guaranteed by adapting the pcm rate constraints when acquiring one or the other stream. > Do you have a requirement that the sample rate must be selected > automatically, instead of being configured with a control panel or with > a CLI tool like amixer? Yes, this is quite crucial as we are attempting a playback system without any local GUI/control panel/CLI interaction, using jackd as audio backend I/O plugin. E.g. starting jackd with jackd -d alsa -d hw1:0 -r <rate> should incorporate switching the samplerate automatically, as this is done with most alsa devices. Given that the ALSA driver API provides for such functionality (snd_xxx_hw_params & snd_xxx_prepare) I thought that such an objective was self-evident. I could think of several other usecases where this would be desired/required. Why _shouldn't_ the application which acquires the DICE device at least be able to dictate the initial samplerate and/or syncsource? What do you (and others) think? Any way we can reach a mutual consent on this? >> I hope that some of these enhancements will make it to your alsa-kernel branch, >> if not even to the 3.x kernel tree (is this being considered for the near future?). > > This branch will land in the kernel when it's reasonably complete. > The plan is for this kernel driver to replace FFADO's streaming > component. If this is the case it would appear that automatical samplerate switching, as is provided by FFADO, is a crucial requirement that should also be transferred to this new kernel driver. >> I noticed that amdtp_out_stream offers support for "dualwire" >> streaming at high rates, which is applied for dice's stream. Has this >> already been considered in FFADO? > > Not by anyone who understands the FFADO internals and has the time. Ok, I'll have a look at it. >> 2.) support DICE capture streams > > I'm working on this right now (modulo the amount of time available). Excellent, we're happy to serve as guinea pigs, whenever you're ready ;-) >> 3.) DICE clock source (CLOCK_SOURCE_*) selectable via ALSA driver API; how could >> this be done? > > If not done by a control panel, this could be done with a mixer control. > This is not yet implemented because any clock source except ARX* > requires synchronization from the capture stream. > >> I have changed the default clock source to INTERNAL, since I was >> experiencing "slips" with ARX1; as far as I know ARX1/2 is reasonably applicable >> only if there is another clock sync master on the Firewire bus. > > The timing of the stream from the computer to the DICE is synthesized > according to the rules in the AMDTP spec (I hope); this was originally > written for devices based on the OXFW970 chip. (It's synchronized to > the FireWire bus clock.) It's possible that DICE devices want to use > a different value for the transfer delay or some other timing parameter; > a stream whose timing is derived from another DICE's capture stream > would have whatever timing is required. > > ARX1 appears to work fine with my DesktopKonnekt6, but I've never looked > at the error counters. Right, hmm, having scanned the AMDTP (IEC 61883-6) in detail raises some confusion now. Up till now I was under the impression that using AVS-SYT as DICE's clock source is only an option to sync firewire bus slave devices to a clock master device on the same bus. After all, the DesktopKonnekt6 manual mentions that in standalone setup INTERNAL syncsource is selected per default and can not be altered. Citing from DICE's users guide (chapter 6: AVS (Audio Video System)): "[...] All nodes on a 1394 network must be synchronized to one clock called the cycle timer, which is determined by the master node on the network. One cycle of the master nodes‘ cycle timer defines a 1394 cycle. At the beginning of each 1394 cycle the master node transmits a clock sync signal that allows all nodes on the 1394 network to be synchronized to the cycle timer. This maintains synchronicity among all the 1394 nodes. Each 1394 node receives the clock sync signal and uses it to update or correct its local timer. However, this clock correction can cause the local timer to jump forward or backward as it is updated by the clock sync signal, which can reduce the performance of the system. The format of a quadlet of audio data passing through the AVS is configurable. The AVS can be configured to be transparent for 32-bit audio data. In this case the data will not be touched as it passes through the AVS. The AVS can also be configured to support the IEC61883-6 (AM824) steaming model. This allows the AVS to either source AM824 labels from another location, or build its own AM824 labels for the 24-bit data. The AVS can take the various label fields (block sync, user bits, channel status bits, etc) that make up each AM824 label, from different sources, and then pack them together into an AM824 label. [...]" Still confused... any enlightenment or recommendation, perhaps from TCAT, on this? Regards, Rolf

2 1

[alsa-devel] [PATCH 0/2] OMAP2+: DMA: fix src/dst position reporting
by Peter Ujfalusi 03 Nov '11

03 Nov '11

Hello, If the user asks for the sDMA current position before the first data has been transmitted (before the first DMA request has been generated), the reported position is not valid: src position: CSAC is uninitialized dst position: CDAC is 0 The return values in both case considered invalid. This sitation can be identified by checking if the CDAC register is 0 (it is initialized to 0 in omap_dam_start call). In this case return the programmed source/destination address. The affected omap_get_dma_src_pos/omap_get_dma_dst_pos functions are used by the audio stack mainly for checking the current position of the audio stream. Regards, Peter --- Peter Ujfalusi (2): OMAP2+: DMA: Workaround for invalid source position OMAP2+: DMA: Workaround for invalid destination position arch/arm/plat-omap/dma.c | 25 ++++++++++++++++++++++--- 1 files changed, 22 insertions(+), 3 deletions(-) -- 1.7.7.1

3 5

[alsa-devel] QasConfig 0.3 released
by Sebastian H. 03 Nov '11

03 Nov '11

QasConfig version 0.3 is now available. QasConfig is a browser for the ALSA configuration tree. Changes: * Changes in /etc/asound.conf or ~/.asoundrc trigger a refresh * Localizations moved to a separate package: qasconfig-l10n * Version code shortened to two numbers instead of three Homepage with more information http://xwmw.org/qasconfig Project page http://sourceforge.net/projects/qasconfig/ Happy configuration browsing! -- Sebastian Holtermann

1 0

[alsa-devel] QasMixer 0.15 released
by Sebastian H. 03 Nov '11

03 Nov '11

QasMixer version 0.15 is now available. QasMixer is an ALSA mixer with a size adaptive Qt GUI. Changes: * New simpler device selection view * New user setting: Mixer device on startup * ALSA configuration view moved to a separate application: QasConfig * Localizations moved to a separate package: qasmixer-l10n * Default fallback translation * Code merges and cleanups * Version code shortened to two numbers instead of three Homepage with more information http://xwmw.org/qasmixer Project page http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] ASoC and regmap trees have moved back to kernel.org
by Mark Brown 03 Nov '11

03 Nov '11

Hi Stephen, Please update my git trees in -next to use kernel.org again: ASoC: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next regmap: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap.git for-next Please leave the regulator tree where it is for now, Liam's kernel.org account will hopefully be being restored shortly so he can take that over again. Thanks, Mark

2 1

[alsa-devel] [PATCH] ASoC: Update git repository URL
by Mark Brown 02 Nov '11

02 Nov '11

Remove the -2.6 from the name. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- MAINTAINERS | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index bbf42cd..63b58dd 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6001,7 +6001,7 @@ F: sound/ 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-2.6.git +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 -- 1.7.7

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[alsa-devel] ALSA firewire kernel branch: snd_dice enhancements
by Rolf Anderegg 02 Nov '11

02 Nov '11

Hi Clemens, hi Stefan, I recently stumbled upon your alsa-kernel branch (firewire-kernel-streaming: http://git.alsa-project.org/?p=alsa-kprivate.git;a=shortlog;h=firewire-kern…) and have finally found the time to play around with it. First of all I was impressed to see that it easily worked in conjunction with our DICE-based devices, even in high samplerate mode (>96kHz), provided that the clockselection of the DICE had been done beforehand via TCAT's drivers. To get samplerate switching working via ALSA's driver API I have made a few changes, especially within the dice_hw_params callback method. Plus, I enhanced the rate constraints for the pcm settings. You can find these changes in the attached patch (applicable to http://git.alsa-project.org/?p=alsa-kprivate.git;a=commit;h=f29bf150b7ba062…). I hope that some of these enhancements will make it to your alsa-kernel branch, if not even to the 3.x kernel tree (is this being considered for the near future?). I am happy to say that this solution will fit our needs for the time being, while we still have not given up on FFADO as a longterm alternative, provided it is stable at all rates. I noticed that amdtp_out_stream offers support for "dualwire" streaming at high rates, which is applied for dice's stream. Has this already been considered in FFADO? I wasn't able to pinpoint the appropriate entities within the FFADO sources yet. In order for snd_dice to make it upstream for generic DICE support there appear to remain a few minor issues: 1.) wait for NOTIFY_CLOCK_ACCEPTED notification bit after writing GLOBAL_CLOCK_SELECT within dice_hw_params. For the time being I introduced a "childish" msleep(500) in order for DICE to potentially switch its ratemode before starting the stream. I didn't quite figure out how to wait/poll for a particular notification. Do you have a hint how this would be done (using hwdep)? 2.) support DICE capture streams 3.) DICE clock source (CLOCK_SOURCE_*) selectable via ALSA driver API; how could this be done? I have changed the default clock source to INTERNAL, since I was experiencing "slips" with ARX1; as far as I know ARX1/2 is reasonably applicable only if there is another clock sync master on the Firewire bus. Thanks again for this cool branch and let us know whether we can be of further assistance. Cheers, Rolf Anderegg, Weiss Engineering Ltd.

2 1

[alsa-devel] Out of date email address for Jaroslav Kysela on wiki
by Andrew Eikum 02 Nov '11

02 Nov '11

I just sent a documentation patch to alsa-devel and received the following bounce. Looks like <perex(a)suse.cz> is no longer valid. Since the wiki page requests users wanting to work on ALSA docs should send an email to Jaroslav Kysela at that address, I had it in the To: field. I'm happy to update the wiki, but I'm not sure what the best address for Jaroslav is these days. Anyone? Andrew ----- Forwarded message from Mail Delivery System <Mailer-Daemon(a)mail.codeweavers.com> ----- Date: Wed, 02 Nov 2011 12:40:58 -0500 From: Mail Delivery System <Mailer-Daemon(a)mail.codeweavers.com> To: aeikum(a)codeweavers.com Subject: Mail delivery failed: returning message to sender This message was created automatically by mail delivery software. A message that you sent could not be delivered to one or more of its recipients. This is a permanent error. The following address(es) failed: perex(a)suse.cz SMTP error from remote mail server after RCPT TO:<perex(a)suse.cz>: host mx1.suse.de [195.135.220.2]: 550 <perex(a)suse.cz>: Recipient address rejected: User unknown in virtual alias table

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[alsa-devel] [alsa-lib PATCH] Remove docs for snd_names_list and snd_names_list_free
by Andrew Eikum 02 Nov '11

02 Nov '11

"Deprecated," to me, doesn't necessarily imply "completely useless," especially if there are complete docs attached. To prevent future confusion, make it clear that these functions are now unimplemented. --- src/names.c | 13 ++----------- 1 files changed, 2 insertions(+), 11 deletions(-)

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[alsa-devel] [PATCH 1/2] ASoC: Sort LM4857 with the CODECs in the Makefile
by Mark Brown 02 Nov '11

02 Nov '11

Having a separate list for amps is a little confusing now the official driver model for them is the same as for other CODECs so let's sort them into the CODEC list, but only do this for those that are actual CODEC drivers so it's easier to remember which ones need updating. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Makefile | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index aaa7fdf..cb83de9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o +snd-soc-lm4857-objs := lm4857.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o @@ -92,7 +93,6 @@ snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-jz4740-codec-objs := jz4740.o # Amp -snd-soc-lm4857-objs := lm4857.o snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-wm2000-objs := wm2000.o @@ -123,6 +123,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o @@ -192,7 +193,6 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp -obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o -- 1.7.7.1

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[alsa-devel] ASoC updates for 3.2
by Mark Brown 02 Nov '11

02 Nov '11

The following changes since commit 5927f94700e860ae27ff24e7f3bc9e4f7b9922eb: ASoC: wm8940: Properly set codec->dapm.bias_level (2011-10-27 09:17:08 +0200) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 The usual connection of small, local fixes to specific drivers. Axel Lin (5): ASoC: wm8940: Fix mask for setting BCLKDIV ASoC: tlv320aic23: Clear TLV320AIC23_MS_MASTER bit for slave mode ASoC: tlv320aic3x: Clear BIT_CLK_MASTER and WORD_CLK_MASTER bits for for slave mode ASoC: wm8711: Fix wrong mask for setting input audio data bit length select ASoC: Leave input audio data bit length settings untouched in wm8711_set_dai_fmt Hong Xu (1): ASoC: WM8904: Set `invert' bit for Capture Switch Mark Brown (4): ASoC: Fix return value of wm5100_gpio_direction_out() ASoC: Ensure WM8962 PLL registers are reset ASoC: Ensure the WM8962 oscillator and PLLs start up disabled ASoC: Ensure we always delay for WM8962 FLL when starting from SYSCLK sound/soc/codecs/tlv320aic23.c | 1 + sound/soc/codecs/tlv320aic3x.c | 1 + sound/soc/codecs/wm5100.c | 12 ++++++++---- sound/soc/codecs/wm8711.c | 4 ++-- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8962.c | 30 ++++++++++++++++++++---------- 7 files changed, 34 insertions(+), 18 deletions(-)

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[alsa-devel] Where to find a guideline for ALSA mixer and control interface?
by Lin, Mengdong 02 Nov '11

02 Nov '11

Could someone tell me where to find a guideline on how to use ALSA mixer & control API? Is reading the source code of amixer my only choice? But I want to have a big picture about these API at first. Many thanks! I found there are many online documents about PCM interface but few about mixer or control interface :( Thanks & Best Regards Mengdong

1 0

[alsa-devel] How to understand capabilities of a mixer control output by "amixer"?
by Lin, Mengdong 02 Nov '11

02 Nov '11

I'm new to ALSA. Could anybody help explain the capabilities values of an ALSA mixer control? - What's the difference between pvolume and pvolume-joined? Only 'Master' control has pvolume-joined, but I cannot see any GUI difference between 'Master' and 'Headerphone' by alsamixer. - What does the penum and cenum mean? What are they designed for? I think "p" means playback and "c" means capture. But it's confusing to see control 'Capture' also has a capability "penum"? Here is output of amixer on my destop: Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 51 [80%] [-13.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 60 [94%] [-4.00dB] [on] Front Right: Playback 60 [94%] [-4.00dB] [on] .... Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [on] Front Right: Capture 0 [0%] [-13.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 9 [29%] [0.00dB] [off] Front Right: Capture 9 [29%] [0.00dB] [off] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Line' Item0: 'Line' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Line' Item0: 'Line' Thanks & Best Regards Mengdong

1 0

[alsa-devel] [PATCH 1/3] ASoC: Convert WM8996 to direct regmap API usage
by Mark Brown 01 Nov '11

01 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 699 +++++++++++++++++++++++++-------------------- 1 files changed, 391 insertions(+), 308 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 645c980..2cea205 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -19,6 +19,7 @@ #include <linux/gcd.h> #include <linux/gpio.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <linux/slab.h> #include <linux/workqueue.h> @@ -114,297 +115,365 @@ WM8996_REGULATOR_EVENT(0) WM8996_REGULATOR_EVENT(1) WM8996_REGULATOR_EVENT(2) -static const u16 wm8996_reg[WM8996_MAX_REGISTER] = { - [WM8996_SOFTWARE_RESET] = 0x8996, - [WM8996_POWER_MANAGEMENT_7] = 0x10, - [WM8996_DAC1_HPOUT1_VOLUME] = 0x88, - [WM8996_DAC2_HPOUT2_VOLUME] = 0x88, - [WM8996_DAC1_LEFT_VOLUME] = 0x2c0, - [WM8996_DAC1_RIGHT_VOLUME] = 0x2c0, - [WM8996_DAC2_LEFT_VOLUME] = 0x2c0, - [WM8996_DAC2_RIGHT_VOLUME] = 0x2c0, - [WM8996_OUTPUT1_LEFT_VOLUME] = 0x80, - [WM8996_OUTPUT1_RIGHT_VOLUME] = 0x80, - [WM8996_OUTPUT2_LEFT_VOLUME] = 0x80, - [WM8996_OUTPUT2_RIGHT_VOLUME] = 0x80, - [WM8996_MICBIAS_1] = 0x39, - [WM8996_MICBIAS_2] = 0x39, - [WM8996_LDO_1] = 0x3, - [WM8996_LDO_2] = 0x13, - [WM8996_ACCESSORY_DETECT_MODE_1] = 0x4, - [WM8996_HEADPHONE_DETECT_1] = 0x20, - [WM8996_MIC_DETECT_1] = 0x7600, - [WM8996_MIC_DETECT_2] = 0xbf, - [WM8996_CHARGE_PUMP_1] = 0x1f25, - [WM8996_CHARGE_PUMP_2] = 0xab19, - [WM8996_DC_SERVO_5] = 0x2a2a, - [WM8996_CONTROL_INTERFACE_1] = 0x8004, - [WM8996_CLOCKING_1] = 0x10, - [WM8996_AIF_RATE] = 0x83, - [WM8996_FLL_CONTROL_4] = 0x5dc0, - [WM8996_FLL_CONTROL_5] = 0xc84, - [WM8996_FLL_EFS_2] = 0x2, - [WM8996_AIF1_TX_LRCLK_1] = 0x80, - [WM8996_AIF1_TX_LRCLK_2] = 0x8, - [WM8996_AIF1_RX_LRCLK_1] = 0x80, - [WM8996_AIF1TX_DATA_CONFIGURATION_1] = 0x1818, - [WM8996_AIF1RX_DATA_CONFIGURATION] = 0x1818, - [WM8996_AIF1TX_TEST] = 0x7, - [WM8996_AIF2_TX_LRCLK_1] = 0x80, - [WM8996_AIF2_TX_LRCLK_2] = 0x8, - [WM8996_AIF2_RX_LRCLK_1] = 0x80, - [WM8996_AIF2TX_DATA_CONFIGURATION_1] = 0x1818, - [WM8996_AIF2RX_DATA_CONFIGURATION] = 0x1818, - [WM8996_AIF2TX_TEST] = 0x1, - [WM8996_DSP1_TX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP1_TX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP1_RX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP1_RX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP1_TX_FILTERS] = 0x2000, - [WM8996_DSP1_RX_FILTERS_1] = 0x200, - [WM8996_DSP1_RX_FILTERS_2] = 0x10, - [WM8996_DSP1_DRC_1] = 0x98, - [WM8996_DSP1_DRC_2] = 0x845, - [WM8996_DSP1_RX_EQ_GAINS_1] = 0x6318, - [WM8996_DSP1_RX_EQ_GAINS_2] = 0x6300, - [WM8996_DSP1_RX_EQ_BAND_1_A] = 0xfca, - [WM8996_DSP1_RX_EQ_BAND_1_B] = 0x400, - [WM8996_DSP1_RX_EQ_BAND_1_PG] = 0xd8, - [WM8996_DSP1_RX_EQ_BAND_2_A] = 0x1eb5, - [WM8996_DSP1_RX_EQ_BAND_2_B] = 0xf145, - [WM8996_DSP1_RX_EQ_BAND_2_C] = 0xb75, - [WM8996_DSP1_RX_EQ_BAND_2_PG] = 0x1c5, - [WM8996_DSP1_RX_EQ_BAND_3_A] = 0x1c58, - [WM8996_DSP1_RX_EQ_BAND_3_B] = 0xf373, - [WM8996_DSP1_RX_EQ_BAND_3_C] = 0xa54, - [WM8996_DSP1_RX_EQ_BAND_3_PG] = 0x558, - [WM8996_DSP1_RX_EQ_BAND_4_A] = 0x168e, - [WM8996_DSP1_RX_EQ_BAND_4_B] = 0xf829, - [WM8996_DSP1_RX_EQ_BAND_4_C] = 0x7ad, - [WM8996_DSP1_RX_EQ_BAND_4_PG] = 0x1103, - [WM8996_DSP1_RX_EQ_BAND_5_A] = 0x564, - [WM8996_DSP1_RX_EQ_BAND_5_B] = 0x559, - [WM8996_DSP1_RX_EQ_BAND_5_PG] = 0x4000, - [WM8996_DSP2_TX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP2_TX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP2_RX_LEFT_VOLUME] = 0xc0, - [WM8996_DSP2_RX_RIGHT_VOLUME] = 0xc0, - [WM8996_DSP2_TX_FILTERS] = 0x2000, - [WM8996_DSP2_RX_FILTERS_1] = 0x200, - [WM8996_DSP2_RX_FILTERS_2] = 0x10, - [WM8996_DSP2_DRC_1] = 0x98, - [WM8996_DSP2_DRC_2] = 0x845, - [WM8996_DSP2_RX_EQ_GAINS_1] = 0x6318, - [WM8996_DSP2_RX_EQ_GAINS_2] = 0x6300, - [WM8996_DSP2_RX_EQ_BAND_1_A] = 0xfca, - [WM8996_DSP2_RX_EQ_BAND_1_B] = 0x400, - [WM8996_DSP2_RX_EQ_BAND_1_PG] = 0xd8, - [WM8996_DSP2_RX_EQ_BAND_2_A] = 0x1eb5, - [WM8996_DSP2_RX_EQ_BAND_2_B] = 0xf145, - [WM8996_DSP2_RX_EQ_BAND_2_C] = 0xb75, - [WM8996_DSP2_RX_EQ_BAND_2_PG] = 0x1c5, - [WM8996_DSP2_RX_EQ_BAND_3_A] = 0x1c58, - [WM8996_DSP2_RX_EQ_BAND_3_B] = 0xf373, - [WM8996_DSP2_RX_EQ_BAND_3_C] = 0xa54, - [WM8996_DSP2_RX_EQ_BAND_3_PG] = 0x558, - [WM8996_DSP2_RX_EQ_BAND_4_A] = 0x168e, - [WM8996_DSP2_RX_EQ_BAND_4_B] = 0xf829, - [WM8996_DSP2_RX_EQ_BAND_4_C] = 0x7ad, - [WM8996_DSP2_RX_EQ_BAND_4_PG] = 0x1103, - [WM8996_DSP2_RX_EQ_BAND_5_A] = 0x564, - [WM8996_DSP2_RX_EQ_BAND_5_B] = 0x559, - [WM8996_DSP2_RX_EQ_BAND_5_PG] = 0x4000, - [WM8996_OVERSAMPLING] = 0xd, - [WM8996_SIDETONE] = 0x1040, - [WM8996_GPIO_1] = 0xa101, - [WM8996_GPIO_2] = 0xa101, - [WM8996_GPIO_3] = 0xa101, - [WM8996_GPIO_4] = 0xa101, - [WM8996_GPIO_5] = 0xa101, - [WM8996_PULL_CONTROL_2] = 0x140, - [WM8996_INTERRUPT_STATUS_1_MASK] = 0x1f, - [WM8996_INTERRUPT_STATUS_2_MASK] = 0x1ecf, - [WM8996_RIGHT_PDM_SPEAKER] = 0x1, - [WM8996_PDM_SPEAKER_MUTE_SEQUENCE] = 0x69, - [WM8996_PDM_SPEAKER_VOLUME] = 0x66, - [WM8996_WRITE_SEQUENCER_0] = 0x1, - [WM8996_WRITE_SEQUENCER_1] = 0x1, - [WM8996_WRITE_SEQUENCER_3] = 0x6, - [WM8996_WRITE_SEQUENCER_4] = 0x40, - [WM8996_WRITE_SEQUENCER_5] = 0x1, - [WM8996_WRITE_SEQUENCER_6] = 0xf, - [WM8996_WRITE_SEQUENCER_7] = 0x6, - [WM8996_WRITE_SEQUENCER_8] = 0x1, - [WM8996_WRITE_SEQUENCER_9] = 0x3, - [WM8996_WRITE_SEQUENCER_10] = 0x104, - [WM8996_WRITE_SEQUENCER_12] = 0x60, - [WM8996_WRITE_SEQUENCER_13] = 0x11, - [WM8996_WRITE_SEQUENCER_14] = 0x401, - [WM8996_WRITE_SEQUENCER_16] = 0x50, - [WM8996_WRITE_SEQUENCER_17] = 0x3, - [WM8996_WRITE_SEQUENCER_18] = 0x100, - [WM8996_WRITE_SEQUENCER_20] = 0x51, - [WM8996_WRITE_SEQUENCER_21] = 0x3, - [WM8996_WRITE_SEQUENCER_22] = 0x104, - [WM8996_WRITE_SEQUENCER_23] = 0xa, - [WM8996_WRITE_SEQUENCER_24] = 0x60, - [WM8996_WRITE_SEQUENCER_25] = 0x3b, - [WM8996_WRITE_SEQUENCER_26] = 0x502, - [WM8996_WRITE_SEQUENCER_27] = 0x100, - [WM8996_WRITE_SEQUENCER_28] = 0x2fff, - [WM8996_WRITE_SEQUENCER_32] = 0x2fff, - [WM8996_WRITE_SEQUENCER_36] = 0x2fff, - [WM8996_WRITE_SEQUENCER_40] = 0x2fff, - [WM8996_WRITE_SEQUENCER_44] = 0x2fff, - [WM8996_WRITE_SEQUENCER_48] = 0x2fff, - [WM8996_WRITE_SEQUENCER_52] = 0x2fff, - [WM8996_WRITE_SEQUENCER_56] = 0x2fff, - [WM8996_WRITE_SEQUENCER_60] = 0x2fff, - [WM8996_WRITE_SEQUENCER_64] = 0x1, - [WM8996_WRITE_SEQUENCER_65] = 0x1, - [WM8996_WRITE_SEQUENCER_67] = 0x6, - [WM8996_WRITE_SEQUENCER_68] = 0x40, - [WM8996_WRITE_SEQUENCER_69] = 0x1, - [WM8996_WRITE_SEQUENCER_70] = 0xf, - [WM8996_WRITE_SEQUENCER_71] = 0x6, - [WM8996_WRITE_SEQUENCER_72] = 0x1, - [WM8996_WRITE_SEQUENCER_73] = 0x3, - [WM8996_WRITE_SEQUENCER_74] = 0x104, - [WM8996_WRITE_SEQUENCER_76] = 0x60, - [WM8996_WRITE_SEQUENCER_77] = 0x11, - [WM8996_WRITE_SEQUENCER_78] = 0x401, - [WM8996_WRITE_SEQUENCER_80] = 0x50, - [WM8996_WRITE_SEQUENCER_81] = 0x3, - [WM8996_WRITE_SEQUENCER_82] = 0x100, - [WM8996_WRITE_SEQUENCER_84] = 0x60, - [WM8996_WRITE_SEQUENCER_85] = 0x3b, - [WM8996_WRITE_SEQUENCER_86] = 0x502, - [WM8996_WRITE_SEQUENCER_87] = 0x100, - [WM8996_WRITE_SEQUENCER_88] = 0x2fff, - [WM8996_WRITE_SEQUENCER_92] = 0x2fff, - [WM8996_WRITE_SEQUENCER_96] = 0x2fff, - [WM8996_WRITE_SEQUENCER_100] = 0x2fff, - [WM8996_WRITE_SEQUENCER_104] = 0x2fff, - [WM8996_WRITE_SEQUENCER_108] = 0x2fff, - [WM8996_WRITE_SEQUENCER_112] = 0x2fff, - [WM8996_WRITE_SEQUENCER_116] = 0x2fff, - [WM8996_WRITE_SEQUENCER_120] = 0x2fff, - [WM8996_WRITE_SEQUENCER_124] = 0x2fff, - [WM8996_WRITE_SEQUENCER_128] = 0x1, - [WM8996_WRITE_SEQUENCER_129] = 0x1, - [WM8996_WRITE_SEQUENCER_131] = 0x6, - [WM8996_WRITE_SEQUENCER_132] = 0x40, - [WM8996_WRITE_SEQUENCER_133] = 0x1, - [WM8996_WRITE_SEQUENCER_134] = 0xf, - [WM8996_WRITE_SEQUENCER_135] = 0x6, - [WM8996_WRITE_SEQUENCER_136] = 0x1, - [WM8996_WRITE_SEQUENCER_137] = 0x3, - [WM8996_WRITE_SEQUENCER_138] = 0x106, - [WM8996_WRITE_SEQUENCER_140] = 0x61, - [WM8996_WRITE_SEQUENCER_141] = 0x11, - [WM8996_WRITE_SEQUENCER_142] = 0x401, - [WM8996_WRITE_SEQUENCER_144] = 0x50, - [WM8996_WRITE_SEQUENCER_145] = 0x3, - [WM8996_WRITE_SEQUENCER_146] = 0x102, - [WM8996_WRITE_SEQUENCER_148] = 0x51, - [WM8996_WRITE_SEQUENCER_149] = 0x3, - [WM8996_WRITE_SEQUENCER_150] = 0x106, - [WM8996_WRITE_SEQUENCER_151] = 0xa, - [WM8996_WRITE_SEQUENCER_152] = 0x61, - [WM8996_WRITE_SEQUENCER_153] = 0x3b, - [WM8996_WRITE_SEQUENCER_154] = 0x502, - [WM8996_WRITE_SEQUENCER_155] = 0x100, - [WM8996_WRITE_SEQUENCER_156] = 0x2fff, - [WM8996_WRITE_SEQUENCER_160] = 0x2fff, - [WM8996_WRITE_SEQUENCER_164] = 0x2fff, - [WM8996_WRITE_SEQUENCER_168] = 0x2fff, - [WM8996_WRITE_SEQUENCER_172] = 0x2fff, - [WM8996_WRITE_SEQUENCER_176] = 0x2fff, - [WM8996_WRITE_SEQUENCER_180] = 0x2fff, - [WM8996_WRITE_SEQUENCER_184] = 0x2fff, - [WM8996_WRITE_SEQUENCER_188] = 0x2fff, - [WM8996_WRITE_SEQUENCER_192] = 0x1, - [WM8996_WRITE_SEQUENCER_193] = 0x1, - [WM8996_WRITE_SEQUENCER_195] = 0x6, - [WM8996_WRITE_SEQUENCER_196] = 0x40, - [WM8996_WRITE_SEQUENCER_197] = 0x1, - [WM8996_WRITE_SEQUENCER_198] = 0xf, - [WM8996_WRITE_SEQUENCER_199] = 0x6, - [WM8996_WRITE_SEQUENCER_200] = 0x1, - [WM8996_WRITE_SEQUENCER_201] = 0x3, - [WM8996_WRITE_SEQUENCER_202] = 0x106, - [WM8996_WRITE_SEQUENCER_204] = 0x61, - [WM8996_WRITE_SEQUENCER_205] = 0x11, - [WM8996_WRITE_SEQUENCER_206] = 0x401, - [WM8996_WRITE_SEQUENCER_208] = 0x50, - [WM8996_WRITE_SEQUENCER_209] = 0x3, - [WM8996_WRITE_SEQUENCER_210] = 0x102, - [WM8996_WRITE_SEQUENCER_212] = 0x61, - [WM8996_WRITE_SEQUENCER_213] = 0x3b, - [WM8996_WRITE_SEQUENCER_214] = 0x502, - [WM8996_WRITE_SEQUENCER_215] = 0x100, - [WM8996_WRITE_SEQUENCER_216] = 0x2fff, - [WM8996_WRITE_SEQUENCER_220] = 0x2fff, - [WM8996_WRITE_SEQUENCER_224] = 0x2fff, - [WM8996_WRITE_SEQUENCER_228] = 0x2fff, - [WM8996_WRITE_SEQUENCER_232] = 0x2fff, - [WM8996_WRITE_SEQUENCER_236] = 0x2fff, - [WM8996_WRITE_SEQUENCER_240] = 0x2fff, - [WM8996_WRITE_SEQUENCER_244] = 0x2fff, - [WM8996_WRITE_SEQUENCER_248] = 0x2fff, - [WM8996_WRITE_SEQUENCER_252] = 0x2fff, - [WM8996_WRITE_SEQUENCER_256] = 0x60, - [WM8996_WRITE_SEQUENCER_258] = 0x601, - [WM8996_WRITE_SEQUENCER_260] = 0x50, - [WM8996_WRITE_SEQUENCER_262] = 0x100, - [WM8996_WRITE_SEQUENCER_264] = 0x1, - [WM8996_WRITE_SEQUENCER_266] = 0x104, - [WM8996_WRITE_SEQUENCER_267] = 0x100, - [WM8996_WRITE_SEQUENCER_268] = 0x2fff, - [WM8996_WRITE_SEQUENCER_272] = 0x2fff, - [WM8996_WRITE_SEQUENCER_276] = 0x2fff, - [WM8996_WRITE_SEQUENCER_280] = 0x2fff, - [WM8996_WRITE_SEQUENCER_284] = 0x2fff, - [WM8996_WRITE_SEQUENCER_288] = 0x2fff, - [WM8996_WRITE_SEQUENCER_292] = 0x2fff, - [WM8996_WRITE_SEQUENCER_296] = 0x2fff, - [WM8996_WRITE_SEQUENCER_300] = 0x2fff, - [WM8996_WRITE_SEQUENCER_304] = 0x2fff, - [WM8996_WRITE_SEQUENCER_308] = 0x2fff, - [WM8996_WRITE_SEQUENCER_312] = 0x2fff, - [WM8996_WRITE_SEQUENCER_316] = 0x2fff, - [WM8996_WRITE_SEQUENCER_320] = 0x61, - [WM8996_WRITE_SEQUENCER_322] = 0x601, - [WM8996_WRITE_SEQUENCER_324] = 0x50, - [WM8996_WRITE_SEQUENCER_326] = 0x102, - [WM8996_WRITE_SEQUENCER_328] = 0x1, - [WM8996_WRITE_SEQUENCER_330] = 0x106, - [WM8996_WRITE_SEQUENCER_331] = 0x100, - [WM8996_WRITE_SEQUENCER_332] = 0x2fff, - [WM8996_WRITE_SEQUENCER_336] = 0x2fff, - [WM8996_WRITE_SEQUENCER_340] = 0x2fff, - [WM8996_WRITE_SEQUENCER_344] = 0x2fff, - [WM8996_WRITE_SEQUENCER_348] = 0x2fff, - [WM8996_WRITE_SEQUENCER_352] = 0x2fff, - [WM8996_WRITE_SEQUENCER_356] = 0x2fff, - [WM8996_WRITE_SEQUENCER_360] = 0x2fff, - [WM8996_WRITE_SEQUENCER_364] = 0x2fff, - [WM8996_WRITE_SEQUENCER_368] = 0x2fff, - [WM8996_WRITE_SEQUENCER_372] = 0x2fff, - [WM8996_WRITE_SEQUENCER_376] = 0x2fff, - [WM8996_WRITE_SEQUENCER_380] = 0x2fff, - [WM8996_WRITE_SEQUENCER_384] = 0x60, - [WM8996_WRITE_SEQUENCER_386] = 0x601, - [WM8996_WRITE_SEQUENCER_388] = 0x61, - [WM8996_WRITE_SEQUENCER_390] = 0x601, - [WM8996_WRITE_SEQUENCER_392] = 0x50, - [WM8996_WRITE_SEQUENCER_394] = 0x300, - [WM8996_WRITE_SEQUENCER_396] = 0x1, - [WM8996_WRITE_SEQUENCER_398] = 0x304, - [WM8996_WRITE_SEQUENCER_400] = 0x40, - [WM8996_WRITE_SEQUENCER_402] = 0xf, - [WM8996_WRITE_SEQUENCER_404] = 0x1, - [WM8996_WRITE_SEQUENCER_407] = 0x100, +static struct reg_default wm8996_reg[] = { + { WM8996_SOFTWARE_RESET, 0x8996 }, + { WM8996_POWER_MANAGEMENT_1, 0x0 }, + { WM8996_POWER_MANAGEMENT_2, 0x0 }, + { WM8996_POWER_MANAGEMENT_3, 0x0 }, + { WM8996_POWER_MANAGEMENT_4, 0x0 }, + { WM8996_POWER_MANAGEMENT_5, 0x0 }, + { WM8996_POWER_MANAGEMENT_6, 0x0 }, + { WM8996_POWER_MANAGEMENT_7, 0x10 }, + { WM8996_POWER_MANAGEMENT_8, 0x0 }, + { WM8996_LEFT_LINE_INPUT_VOLUME, 0x0 }, + { WM8996_RIGHT_LINE_INPUT_VOLUME, 0x0 }, + { WM8996_LINE_INPUT_CONTROL, 0x0 }, + { WM8996_DAC1_HPOUT1_VOLUME, 0x88 }, + { WM8996_DAC2_HPOUT2_VOLUME, 0x88 }, + { WM8996_DAC1_LEFT_VOLUME, 0x2c0 }, + { WM8996_DAC1_RIGHT_VOLUME, 0x2c0 }, + { WM8996_DAC2_LEFT_VOLUME, 0x2c0 }, + { WM8996_DAC2_RIGHT_VOLUME, 0x2c0 }, + { WM8996_OUTPUT1_LEFT_VOLUME, 0x80 }, + { WM8996_OUTPUT1_RIGHT_VOLUME, 0x80 }, + { WM8996_OUTPUT2_LEFT_VOLUME, 0x80 }, + { WM8996_OUTPUT2_RIGHT_VOLUME, 0x80 }, + { WM8996_MICBIAS_1, 0x39 }, + { WM8996_MICBIAS_2, 0x39 }, + { WM8996_LDO_1, 0x3 }, + { WM8996_LDO_2, 0x13 }, + { WM8996_ACCESSORY_DETECT_MODE_1, 0x4 }, + { WM8996_ACCESSORY_DETECT_MODE_2, 0x0 }, + { WM8996_HEADPHONE_DETECT_1, 0x20 }, + { WM8996_HEADPHONE_DETECT_2, 0x0 }, + { WM8996_MIC_DETECT_1, 0x7600 }, + { WM8996_MIC_DETECT_2, 0xbf }, + { WM8996_CHARGE_PUMP_1, 0x1f25 }, + { WM8996_CHARGE_PUMP_2, 0xab19 }, + { WM8996_DC_SERVO_1, 0x0 }, + { WM8996_DC_SERVO_2, 0x0 }, + { WM8996_DC_SERVO_3, 0x0 }, + { WM8996_DC_SERVO_5, 0x2a2a }, + { WM8996_DC_SERVO_6, 0x0 }, + { WM8996_DC_SERVO_7, 0x0 }, + { WM8996_ANALOGUE_HP_1, 0x0 }, + { WM8996_ANALOGUE_HP_2, 0x0 }, + { WM8996_CONTROL_INTERFACE_1, 0x8004 }, + { WM8996_WRITE_SEQUENCER_CTRL_1, 0x0 }, + { WM8996_WRITE_SEQUENCER_CTRL_2, 0x0 }, + { WM8996_AIF_CLOCKING_1, 0x0 }, + { WM8996_AIF_CLOCKING_2, 0x0 }, + { WM8996_CLOCKING_1, 0x10 }, + { WM8996_CLOCKING_2, 0x0 }, + { WM8996_AIF_RATE, 0x83 }, + { WM8996_FLL_CONTROL_1, 0x0 }, + { WM8996_FLL_CONTROL_2, 0x0 }, + { WM8996_FLL_CONTROL_3, 0x0 }, + { WM8996_FLL_CONTROL_4, 0x5dc0 }, + { WM8996_FLL_CONTROL_5, 0xc84 }, + { WM8996_FLL_EFS_1, 0x0 }, + { WM8996_FLL_EFS_2, 0x2 }, + { WM8996_AIF1_CONTROL, 0x0 }, + { WM8996_AIF1_BCLK, 0x0 }, + { WM8996_AIF1_TX_LRCLK_1, 0x80 }, + { WM8996_AIF1_TX_LRCLK_2, 0x8 }, + { WM8996_AIF1_RX_LRCLK_1, 0x80 }, + { WM8996_AIF1_RX_LRCLK_2, 0x0 }, + { WM8996_AIF1TX_DATA_CONFIGURATION_1, 0x1818 }, + { WM8996_AIF1TX_DATA_CONFIGURATION_2, 0 }, + { WM8996_AIF1RX_DATA_CONFIGURATION, 0x1818 }, + { WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, 0x0 }, + { WM8996_AIF1RX_MONO_CONFIGURATION, 0x0 }, + { WM8996_AIF1TX_TEST, 0x7 }, + { WM8996_AIF2_CONTROL, 0x0 }, + { WM8996_AIF2_BCLK, 0x0 }, + { WM8996_AIF2_TX_LRCLK_1, 0x80 }, + { WM8996_AIF2_TX_LRCLK_2, 0x8 }, + { WM8996_AIF2_RX_LRCLK_1, 0x80 }, + { WM8996_AIF2_RX_LRCLK_2, 0x0 }, + { WM8996_AIF2TX_DATA_CONFIGURATION_1, 0x1818 }, + { WM8996_AIF2RX_DATA_CONFIGURATION, 0x1818 }, + { WM8996_AIF2RX_DATA_CONFIGURATION, 0x0 }, + { WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF2TX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, 0x0 }, + { WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, 0x0 }, + { WM8996_AIF2RX_MONO_CONFIGURATION, 0x0 }, + { WM8996_AIF2TX_TEST, 0x1 }, + { WM8996_DSP1_TX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP1_TX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP1_RX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP1_RX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP1_TX_FILTERS, 0x2000 }, + { WM8996_DSP1_RX_FILTERS_1, 0x200 }, + { WM8996_DSP1_RX_FILTERS_2, 0x10 }, + { WM8996_DSP1_DRC_1, 0x98 }, + { WM8996_DSP1_DRC_2, 0x845 }, + { WM8996_DSP1_RX_EQ_GAINS_1, 0x6318 }, + { WM8996_DSP1_RX_EQ_GAINS_2, 0x6300 }, + { WM8996_DSP1_RX_EQ_BAND_1_A, 0xfca }, + { WM8996_DSP1_RX_EQ_BAND_1_B, 0x400 }, + { WM8996_DSP1_RX_EQ_BAND_1_PG, 0xd8 }, + { WM8996_DSP1_RX_EQ_BAND_2_A, 0x1eb5 }, + { WM8996_DSP1_RX_EQ_BAND_2_B, 0xf145 }, + { WM8996_DSP1_RX_EQ_BAND_2_C, 0xb75 }, + { WM8996_DSP1_RX_EQ_BAND_2_PG, 0x1c5 }, + { WM8996_DSP1_RX_EQ_BAND_3_A, 0x1c58 }, + { WM8996_DSP1_RX_EQ_BAND_3_B, 0xf373 }, + { WM8996_DSP1_RX_EQ_BAND_3_C, 0xa54 }, + { WM8996_DSP1_RX_EQ_BAND_3_PG, 0x558 }, + { WM8996_DSP1_RX_EQ_BAND_4_A, 0x168e }, + { WM8996_DSP1_RX_EQ_BAND_4_B, 0xf829 }, + { WM8996_DSP1_RX_EQ_BAND_4_C, 0x7ad }, + { WM8996_DSP1_RX_EQ_BAND_4_PG, 0x1103 }, + { WM8996_DSP1_RX_EQ_BAND_5_A, 0x564 }, + { WM8996_DSP1_RX_EQ_BAND_5_B, 0x559 }, + { WM8996_DSP1_RX_EQ_BAND_5_PG, 0x4000 }, + { WM8996_DSP2_TX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP2_TX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP2_RX_LEFT_VOLUME, 0xc0 }, + { WM8996_DSP2_RX_RIGHT_VOLUME, 0xc0 }, + { WM8996_DSP2_TX_FILTERS, 0x2000 }, + { WM8996_DSP2_RX_FILTERS_1, 0x200 }, + { WM8996_DSP2_RX_FILTERS_2, 0x10 }, + { WM8996_DSP2_DRC_1, 0x98 }, + { WM8996_DSP2_DRC_2, 0x845 }, + { WM8996_DSP2_RX_EQ_GAINS_1, 0x6318 }, + { WM8996_DSP2_RX_EQ_GAINS_2, 0x6300 }, + { WM8996_DSP2_RX_EQ_BAND_1_A, 0xfca }, + { WM8996_DSP2_RX_EQ_BAND_1_B, 0x400 }, + { WM8996_DSP2_RX_EQ_BAND_1_PG, 0xd8 }, + { WM8996_DSP2_RX_EQ_BAND_2_A, 0x1eb5 }, + { WM8996_DSP2_RX_EQ_BAND_2_B, 0xf145 }, + { WM8996_DSP2_RX_EQ_BAND_2_C, 0xb75 }, + { WM8996_DSP2_RX_EQ_BAND_2_PG, 0x1c5 }, + { WM8996_DSP2_RX_EQ_BAND_3_A, 0x1c58 }, + { WM8996_DSP2_RX_EQ_BAND_3_B, 0xf373 }, + { WM8996_DSP2_RX_EQ_BAND_3_C, 0xa54 }, + { WM8996_DSP2_RX_EQ_BAND_3_PG, 0x558 }, + { WM8996_DSP2_RX_EQ_BAND_4_A, 0x168e }, + { WM8996_DSP2_RX_EQ_BAND_4_B, 0xf829 }, + { WM8996_DSP2_RX_EQ_BAND_4_C, 0x7ad }, + { WM8996_DSP2_RX_EQ_BAND_4_PG, 0x1103 }, + { WM8996_DSP2_RX_EQ_BAND_5_A, 0x564 }, + { WM8996_DSP2_RX_EQ_BAND_5_B, 0x559 }, + { WM8996_DSP2_RX_EQ_BAND_5_PG, 0x4000 }, + { WM8996_DAC1_MIXER_VOLUMES, 0x0 }, + { WM8996_DAC1_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DAC1_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DAC2_MIXER_VOLUMES, 0x0 }, + { WM8996_DAC2_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DAC2_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP1_TX_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP1_TX_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP2_TX_LEFT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP2_TX_RIGHT_MIXER_ROUTING, 0x0 }, + { WM8996_DSP_TX_MIXER_SELECT, 0x0 }, + { WM8996_DAC_SOFTMUTE, 0x0 }, + { WM8996_OVERSAMPLING, 0xd }, + { WM8996_SIDETONE, 0x1040 }, + { WM8996_GPIO_1, 0xa101 }, + { WM8996_GPIO_2, 0xa101 }, + { WM8996_GPIO_3, 0xa101 }, + { WM8996_GPIO_4, 0xa101 }, + { WM8996_GPIO_5, 0xa101 }, + { WM8996_PULL_CONTROL_1, 0x0 }, + { WM8996_PULL_CONTROL_2, 0x140 }, + { WM8996_INTERRUPT_STATUS_1_MASK, 0x1f }, + { WM8996_INTERRUPT_STATUS_2_MASK, 0x1ecf }, + { WM8996_LEFT_PDM_SPEAKER, 0x0 }, + { WM8996_RIGHT_PDM_SPEAKER, 0x1 }, + { WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 }, + { WM8996_PDM_SPEAKER_VOLUME, 0x66 }, + { WM8996_WRITE_SEQUENCER_0, 0x1 }, + { WM8996_WRITE_SEQUENCER_1, 0x1 }, + { WM8996_WRITE_SEQUENCER_3, 0x6 }, + { WM8996_WRITE_SEQUENCER_4, 0x40 }, + { WM8996_WRITE_SEQUENCER_5, 0x1 }, + { WM8996_WRITE_SEQUENCER_6, 0xf }, + { WM8996_WRITE_SEQUENCER_7, 0x6 }, + { WM8996_WRITE_SEQUENCER_8, 0x1 }, + { WM8996_WRITE_SEQUENCER_9, 0x3 }, + { WM8996_WRITE_SEQUENCER_10, 0x104 }, + { WM8996_WRITE_SEQUENCER_12, 0x60 }, + { WM8996_WRITE_SEQUENCER_13, 0x11 }, + { WM8996_WRITE_SEQUENCER_14, 0x401 }, + { WM8996_WRITE_SEQUENCER_16, 0x50 }, + { WM8996_WRITE_SEQUENCER_17, 0x3 }, + { WM8996_WRITE_SEQUENCER_18, 0x100 }, + { WM8996_WRITE_SEQUENCER_20, 0x51 }, + { WM8996_WRITE_SEQUENCER_21, 0x3 }, + { WM8996_WRITE_SEQUENCER_22, 0x104 }, + { WM8996_WRITE_SEQUENCER_23, 0xa }, + { WM8996_WRITE_SEQUENCER_24, 0x60 }, + { WM8996_WRITE_SEQUENCER_25, 0x3b }, + { WM8996_WRITE_SEQUENCER_26, 0x502 }, + { WM8996_WRITE_SEQUENCER_27, 0x100 }, + { WM8996_WRITE_SEQUENCER_28, 0x2fff }, + { WM8996_WRITE_SEQUENCER_32, 0x2fff }, + { WM8996_WRITE_SEQUENCER_36, 0x2fff }, + { WM8996_WRITE_SEQUENCER_40, 0x2fff }, + { WM8996_WRITE_SEQUENCER_44, 0x2fff }, + { WM8996_WRITE_SEQUENCER_48, 0x2fff }, + { WM8996_WRITE_SEQUENCER_52, 0x2fff }, + { WM8996_WRITE_SEQUENCER_56, 0x2fff }, + { WM8996_WRITE_SEQUENCER_60, 0x2fff }, + { WM8996_WRITE_SEQUENCER_64, 0x1 }, + { WM8996_WRITE_SEQUENCER_65, 0x1 }, + { WM8996_WRITE_SEQUENCER_67, 0x6 }, + { WM8996_WRITE_SEQUENCER_68, 0x40 }, + { WM8996_WRITE_SEQUENCER_69, 0x1 }, + { WM8996_WRITE_SEQUENCER_70, 0xf }, + { WM8996_WRITE_SEQUENCER_71, 0x6 }, + { WM8996_WRITE_SEQUENCER_72, 0x1 }, + { WM8996_WRITE_SEQUENCER_73, 0x3 }, + { WM8996_WRITE_SEQUENCER_74, 0x104 }, + { WM8996_WRITE_SEQUENCER_76, 0x60 }, + { WM8996_WRITE_SEQUENCER_77, 0x11 }, + { WM8996_WRITE_SEQUENCER_78, 0x401 }, + { WM8996_WRITE_SEQUENCER_80, 0x50 }, + { WM8996_WRITE_SEQUENCER_81, 0x3 }, + { WM8996_WRITE_SEQUENCER_82, 0x100 }, + { WM8996_WRITE_SEQUENCER_84, 0x60 }, + { WM8996_WRITE_SEQUENCER_85, 0x3b }, + { WM8996_WRITE_SEQUENCER_86, 0x502 }, + { WM8996_WRITE_SEQUENCER_87, 0x100 }, + { WM8996_WRITE_SEQUENCER_88, 0x2fff }, + { WM8996_WRITE_SEQUENCER_92, 0x2fff }, + { WM8996_WRITE_SEQUENCER_96, 0x2fff }, + { WM8996_WRITE_SEQUENCER_100, 0x2fff }, + { WM8996_WRITE_SEQUENCER_104, 0x2fff }, + { WM8996_WRITE_SEQUENCER_108, 0x2fff }, + { WM8996_WRITE_SEQUENCER_112, 0x2fff }, + { WM8996_WRITE_SEQUENCER_116, 0x2fff }, + { WM8996_WRITE_SEQUENCER_120, 0x2fff }, + { WM8996_WRITE_SEQUENCER_124, 0x2fff }, + { WM8996_WRITE_SEQUENCER_128, 0x1 }, + { WM8996_WRITE_SEQUENCER_129, 0x1 }, + { WM8996_WRITE_SEQUENCER_131, 0x6 }, + { WM8996_WRITE_SEQUENCER_132, 0x40 }, + { WM8996_WRITE_SEQUENCER_133, 0x1 }, + { WM8996_WRITE_SEQUENCER_134, 0xf }, + { WM8996_WRITE_SEQUENCER_135, 0x6 }, + { WM8996_WRITE_SEQUENCER_136, 0x1 }, + { WM8996_WRITE_SEQUENCER_137, 0x3 }, + { WM8996_WRITE_SEQUENCER_138, 0x106 }, + { WM8996_WRITE_SEQUENCER_140, 0x61 }, + { WM8996_WRITE_SEQUENCER_141, 0x11 }, + { WM8996_WRITE_SEQUENCER_142, 0x401 }, + { WM8996_WRITE_SEQUENCER_144, 0x50 }, + { WM8996_WRITE_SEQUENCER_145, 0x3 }, + { WM8996_WRITE_SEQUENCER_146, 0x102 }, + { WM8996_WRITE_SEQUENCER_148, 0x51 }, + { WM8996_WRITE_SEQUENCER_149, 0x3 }, + { WM8996_WRITE_SEQUENCER_150, 0x106 }, + { WM8996_WRITE_SEQUENCER_151, 0xa }, + { WM8996_WRITE_SEQUENCER_152, 0x61 }, + { WM8996_WRITE_SEQUENCER_153, 0x3b }, + { WM8996_WRITE_SEQUENCER_154, 0x502 }, + { WM8996_WRITE_SEQUENCER_155, 0x100 }, + { WM8996_WRITE_SEQUENCER_156, 0x2fff }, + { WM8996_WRITE_SEQUENCER_160, 0x2fff }, + { WM8996_WRITE_SEQUENCER_164, 0x2fff }, + { WM8996_WRITE_SEQUENCER_168, 0x2fff }, + { WM8996_WRITE_SEQUENCER_172, 0x2fff }, + { WM8996_WRITE_SEQUENCER_176, 0x2fff }, + { WM8996_WRITE_SEQUENCER_180, 0x2fff }, + { WM8996_WRITE_SEQUENCER_184, 0x2fff }, + { WM8996_WRITE_SEQUENCER_188, 0x2fff }, + { WM8996_WRITE_SEQUENCER_192, 0x1 }, + { WM8996_WRITE_SEQUENCER_193, 0x1 }, + { WM8996_WRITE_SEQUENCER_195, 0x6 }, + { WM8996_WRITE_SEQUENCER_196, 0x40 }, + { WM8996_WRITE_SEQUENCER_197, 0x1 }, + { WM8996_WRITE_SEQUENCER_198, 0xf }, + { WM8996_WRITE_SEQUENCER_199, 0x6 }, + { WM8996_WRITE_SEQUENCER_200, 0x1 }, + { WM8996_WRITE_SEQUENCER_201, 0x3 }, + { WM8996_WRITE_SEQUENCER_202, 0x106 }, + { WM8996_WRITE_SEQUENCER_204, 0x61 }, + { WM8996_WRITE_SEQUENCER_205, 0x11 }, + { WM8996_WRITE_SEQUENCER_206, 0x401 }, + { WM8996_WRITE_SEQUENCER_208, 0x50 }, + { WM8996_WRITE_SEQUENCER_209, 0x3 }, + { WM8996_WRITE_SEQUENCER_210, 0x102 }, + { WM8996_WRITE_SEQUENCER_212, 0x61 }, + { WM8996_WRITE_SEQUENCER_213, 0x3b }, + { WM8996_WRITE_SEQUENCER_214, 0x502 }, + { WM8996_WRITE_SEQUENCER_215, 0x100 }, + { WM8996_WRITE_SEQUENCER_216, 0x2fff }, + { WM8996_WRITE_SEQUENCER_220, 0x2fff }, + { WM8996_WRITE_SEQUENCER_224, 0x2fff }, + { WM8996_WRITE_SEQUENCER_228, 0x2fff }, + { WM8996_WRITE_SEQUENCER_232, 0x2fff }, + { WM8996_WRITE_SEQUENCER_236, 0x2fff }, + { WM8996_WRITE_SEQUENCER_240, 0x2fff }, + { WM8996_WRITE_SEQUENCER_244, 0x2fff }, + { WM8996_WRITE_SEQUENCER_248, 0x2fff }, + { WM8996_WRITE_SEQUENCER_252, 0x2fff }, + { WM8996_WRITE_SEQUENCER_256, 0x60 }, + { WM8996_WRITE_SEQUENCER_258, 0x601 }, + { WM8996_WRITE_SEQUENCER_260, 0x50 }, + { WM8996_WRITE_SEQUENCER_262, 0x100 }, + { WM8996_WRITE_SEQUENCER_264, 0x1 }, + { WM8996_WRITE_SEQUENCER_266, 0x104 }, + { WM8996_WRITE_SEQUENCER_267, 0x100 }, + { WM8996_WRITE_SEQUENCER_268, 0x2fff }, + { WM8996_WRITE_SEQUENCER_272, 0x2fff }, + { WM8996_WRITE_SEQUENCER_276, 0x2fff }, + { WM8996_WRITE_SEQUENCER_280, 0x2fff }, + { WM8996_WRITE_SEQUENCER_284, 0x2fff }, + { WM8996_WRITE_SEQUENCER_288, 0x2fff }, + { WM8996_WRITE_SEQUENCER_292, 0x2fff }, + { WM8996_WRITE_SEQUENCER_296, 0x2fff }, + { WM8996_WRITE_SEQUENCER_300, 0x2fff }, + { WM8996_WRITE_SEQUENCER_304, 0x2fff }, + { WM8996_WRITE_SEQUENCER_308, 0x2fff }, + { WM8996_WRITE_SEQUENCER_312, 0x2fff }, + { WM8996_WRITE_SEQUENCER_316, 0x2fff }, + { WM8996_WRITE_SEQUENCER_320, 0x61 }, + { WM8996_WRITE_SEQUENCER_322, 0x601 }, + { WM8996_WRITE_SEQUENCER_324, 0x50 }, + { WM8996_WRITE_SEQUENCER_326, 0x102 }, + { WM8996_WRITE_SEQUENCER_328, 0x1 }, + { WM8996_WRITE_SEQUENCER_330, 0x106 }, + { WM8996_WRITE_SEQUENCER_331, 0x100 }, + { WM8996_WRITE_SEQUENCER_332, 0x2fff }, + { WM8996_WRITE_SEQUENCER_336, 0x2fff }, + { WM8996_WRITE_SEQUENCER_340, 0x2fff }, + { WM8996_WRITE_SEQUENCER_344, 0x2fff }, + { WM8996_WRITE_SEQUENCER_348, 0x2fff }, + { WM8996_WRITE_SEQUENCER_352, 0x2fff }, + { WM8996_WRITE_SEQUENCER_356, 0x2fff }, + { WM8996_WRITE_SEQUENCER_360, 0x2fff }, + { WM8996_WRITE_SEQUENCER_364, 0x2fff }, + { WM8996_WRITE_SEQUENCER_368, 0x2fff }, + { WM8996_WRITE_SEQUENCER_372, 0x2fff }, + { WM8996_WRITE_SEQUENCER_376, 0x2fff }, + { WM8996_WRITE_SEQUENCER_380, 0x2fff }, + { WM8996_WRITE_SEQUENCER_384, 0x60 }, + { WM8996_WRITE_SEQUENCER_386, 0x601 }, + { WM8996_WRITE_SEQUENCER_388, 0x61 }, + { WM8996_WRITE_SEQUENCER_390, 0x601 }, + { WM8996_WRITE_SEQUENCER_392, 0x50 }, + { WM8996_WRITE_SEQUENCER_394, 0x300 }, + { WM8996_WRITE_SEQUENCER_396, 0x1 }, + { WM8996_WRITE_SEQUENCER_398, 0x304 }, + { WM8996_WRITE_SEQUENCER_400, 0x40 }, + { WM8996_WRITE_SEQUENCER_402, 0xf }, + { WM8996_WRITE_SEQUENCER_404, 0x1 }, + { WM8996_WRITE_SEQUENCER_407, 0x100 }, }; static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0); @@ -1413,8 +1482,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "SPKDAT", NULL, "SPKR PGA" }, }; -static int wm8996_readable_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8996_readable_register(struct device *dev, unsigned int reg) { /* Due to the sparseness of the register map the compiler * output from an explicit switch statement ends up being much @@ -1621,8 +1689,7 @@ static int wm8996_readable_register(struct snd_soc_codec *codec, } } -static int wm8996_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool wm8996_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8996_SOFTWARE_RESET: @@ -1723,13 +1790,13 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, msleep(5); } - codec->cache_only = false; - snd_soc_cache_sync(codec); + regcache_cache_only(codec->control_data, false); + regcache_sync(codec->control_data); } break; case SND_SOC_BIAS_OFF: - codec->cache_only = true; + regcache_cache_only(codec->control_data, true); if (wm8996->pdata.ldo_ena >= 0) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), @@ -2692,6 +2759,18 @@ static void wm8996_retune_mobile_pdata(struct snd_soc_codec *codec) "Failed to add ReTune Mobile controls: %d\n", ret); } +static const struct regmap_config wm8996_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = WM8996_MAX_REGISTER, + .reg_defaults = wm8996_reg, + .num_reg_defaults = ARRAY_SIZE(wm8996_reg), + .volatile_reg = wm8996_volatile_register, + .readable_reg = wm8996_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + static int wm8996_probe(struct snd_soc_codec *codec) { int ret; @@ -2707,10 +2786,17 @@ static int wm8996_probe(struct snd_soc_codec *codec) dapm->idle_bias_off = true; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + codec->control_data = regmap_init_i2c(i2c, &wm8996_regmap); + if (IS_ERR(codec->control_data)) { + ret = PTR_ERR(codec->control_data); + dev_err(codec->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err; + goto err_regmap; } for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) @@ -2720,7 +2806,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) wm8996->supplies); if (ret != 0) { dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err; + goto err_regmap; } wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; @@ -2788,7 +2874,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) } } - codec->cache_only = true; + regcache_cache_only(codec->control_data, true); /* Apply platform data settings */ snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, @@ -2996,6 +3082,8 @@ err_cpvdd: regulator_put(wm8996->cpvdd); err_get: regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); +err_regmap: + regmap_exit(codec->control_data); err: return ret; } @@ -3019,6 +3107,7 @@ static int wm8996_remove(struct snd_soc_codec *codec) &wm8996->disable_nb[i]); regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + regmap_exit(codec->control_data); return 0; } @@ -3028,12 +3117,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .remove = wm8996_remove, .set_bias_level = wm8996_set_bias_level, .seq_notifier = wm8996_seq_notifier, - .reg_cache_size = WM8996_MAX_REGISTER + 1, - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8996_reg, - .volatile_register = wm8996_volatile_register, - .readable_register = wm8996_readable_register, - .compress_type = SND_SOC_RBTREE_COMPRESSION, .controls = wm8996_snd_controls, .num_controls = ARRAY_SIZE(wm8996_snd_controls), .dapm_widgets = wm8996_dapm_widgets, -- 1.7.7.1

1 2

Re: [alsa-devel] [PATCH] ASoC don't clobber platform DMA driver ops
by Mark Brown 01 Nov '11

01 Nov '11

On Tue, Nov 01, 2011 at 06:57:47PM +0000, Tull Alan-R80115 wrote: On public Linux lists you should never top post (so people have context and know what you're talking about) and you should fix your mail client to word wrap within paragraphs. > I originally started down the path you are describing, but quickly ran > into problems. I started by adding a soc_pcm_copy and soc_pcm_mmap > and found that I needed to pull code from ALSA core in to handle the > case where copy or mmap was NULL for a given platform driver. In that case your changelog needs to explain this and I'd still use a function for those ops that don't have stubs in the core. Things like ioctl() could usefully be routed to all the drivers, for example.

1 0

Re: [alsa-devel] [PATCH] ASoC don't clobber platform DMA driver ops
by Mark Brown 01 Nov '11

01 Nov '11

On Tue, Nov 01, 2011 at 06:46:52PM +0000, Tull Alan-R80115 wrote: > This patch creates and populates a ops struct for each call to soc_pcm_new. This isn't very idiomatic - a much more idiomatic solution would be to add new indirection functions which do the lookups. While it's probably not going to make much difference for most things there's operations like ioctl() which definitely don't look at all DMA specific in there.

1 0

[alsa-devel] [PATCH] ASoC don't clobber platform DMA driver ops
by Alan Tull 01 Nov '11

01 Nov '11

Previously, only one static struct for ops existed for all platform DMA drivers to share. Half of the ops are shared functions and the other half are initialized to point directly to ops in the platform driver. This creates problems where each time soc_new_pcm is called, the new platform driver's ops would overwrite a subset of the ops. This patch creates and populates a ops struct for each call to soc_pcm_new. Signed-off-by: Alan Tull <alan.tull(a)freescale.com> --- sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 48 ++++++++++++++++++++++++++++-------------------- 2 files changed, 29 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b085d8e..1b48fc6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1547,6 +1547,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_free(&card->dapm); + kfree(card->rtd->ops); kfree(card->rtd); snd_card_free(card->snd_card); return 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2879c88..091c074 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -567,17 +567,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -586,9 +575,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; + struct snd_pcm_ops *soc_pcm_ops; char new_name[64]; int ret = 0, playback = 0, capture = 0; + /* ASoC PCM operations */ + soc_pcm_ops = kzalloc(sizeof(struct snd_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + printk(KERN_ERR "asoc: can't create pcm ops for codec %s\n", codec->name); + return -ENOMEM; + } + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -603,34 +600,45 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) num, playback, capture, &pcm); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + kfree(soc_pcm_ops); return ret; } /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_codec_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->pointer = soc_pcm_pointer; + soc_pcm_ops->trigger = soc_pcm_trigger; + rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } + rtd->ops = soc_pcm_ops; + if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { pr_err("asoc: platform pcm constructor failed\n"); + kfree(rtd->ops); return ret; } } -- 1.6.0.4

1 0

[alsa-devel] [PATCH 1/7] ASoC: Ensure WM8962 PLL registers are reset
by Mark Brown 01 Nov '11

01 Nov '11

The WM8962 has a separate software reset for the PLL registers. Ensure that these are reset also on startup. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8d8b06c..9e2d5b0 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1963,7 +1963,13 @@ static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int re static int wm8962_reset(struct snd_soc_codec *codec) { - return snd_soc_write(codec, WM8962_SOFTWARE_RESET, 0x6243); + int ret; + + ret = snd_soc_write(codec, WM8962_SOFTWARE_RESET, 0x6243); + if (ret != 0) + return ret; + + return snd_soc_write(codec, WM8962_PLL_SOFTWARE_RESET, 0); } static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0); -- 1.7.7.1

1 6

Re: [alsa-devel] [PATCH] asoc: tegra: wm8903: Simplify pin disconnect
by Mark Brown 01 Nov '11

01 Nov '11

On Tue, Nov 01, 2011 at 09:26:11AM -0700, Taylor Hutt wrote: > (Note that this patch fails the kernel patch checker because > the 'M' macro produces an error that it's value should be enclosed > in parenthesis; this is not actually possible for this use & expansion.) So, this whole patch doesn't look like a Linux patch in either the code itself or the way it's presented. I remember having similar problems before with patches you've submitted but this is even further off the mark than previously. I did start replying in detail but there's so many really basic issues from the coding and patch submission style level up that it was taking too long. In general whenever you're modifying code (kernel or otherwise) if what you're doing doesn't visually resemble the rest of the system there's probably a problem.

1 0

Re: [alsa-devel] [PATCH] asoc: tegra: wm8903: Simplify pin disconnect
by Stephen Warren 01 Nov '11

01 Nov '11

Taylor Hutt wrote at Tuesday, November 01, 2011 10:26 AM: > (Note that this patch fails the kernel patch checker because > the 'M' macro produces an error that it's value should be enclosed > in parenthesis; this is not actually possible for this use & expansion.) > > Detail > > This change is a simplification, both in implementation and for > reasoning about whcih pins are connected and disconnected. > > The impetus for the change was the addition of new boards, and the > difficulty in reasoning about the previous code to disconnect pins. > > Now, unless a connection is specified for the current board, the pin > will be disconnected. > > A follow-on change will add the 'asymptote' board. > > Testing > > Visual inspection from original code. > Built kernel. As I mentioned in the ChromeOS review, I don't like this approach. The main issue is that all the connectivity information is already present in the ${machine}_audio_maps[] arrays, and this change duplicates it all in the data-structures inside tegra_wm8903_disconnect_pins(). Instead, I believe a function should be added to the ASoC core that reads the machine's dapm routes, and does the same thing based on that. Mark Brown pointed out that this should be an optional utility function that machine drivers should call, since applying it automatically might cause issues for some other machines. The second issue here is that the new code calls machine_is_ventana() and the equivalent for other machines, and some of those functions will go away next time Russell updates mach-types and removes entries for boards that aren't supported with an explicit board file. -- nvpublic

1 0

Re: [alsa-devel] [PATCH] ASoc : Tegra : Add audio map for Asymptote
by Stephen Warren 01 Nov '11

01 Nov '11

Taylor Hutt wrote at Tuesday, November 01, 2011 10:33 AM: > Details > > Add pin connections and audio map for the Asymptote board. > > Testing > > Visual inspection with Chromium OS 2.6.38 code. > Built kernel > Built chromium os image > Booted image on asymptote > Booted image on kaen Presumably all that testing was within the chromeos-3.0 kernel, not mainline; asymptote isn't supported in mainline. This patch can't be applied for a few reasons: a) Asymptote isn't in mainline's arch/arm/tools/mach-types. This means that function machine_is_asymptote() doesn't exists, and this code can't compile. b) Even if the machine were in mach-types, there's no board file for asymptote, so Russell King's automated mach-types cleanup script would flag asymptote for removal and eventually it'd go away. This is the same reason we recently removed Ventana support from tegra_wm8903.c. I'm not sure how this will work once we have device-tree-only boards; I guess we need to remove all the machine_is_foo() checks from the code and parameterize everything from DT instead. -- nvpublic

1 0

[alsa-devel] sound git tree back to kernel.org
by Takashi Iwai 01 Nov '11

01 Nov '11

Hi, as I got an account again in kernel.org, I moved sound git tree back to kernel.org. But at this time, I dropped -2.6 suffix, by obvious reason. The new URL is: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git The alsa-driver-build.git and unstable trees are also back there, too. The github host will be used for a while as a mirror, but will be removed sometime later. Currently, the daily tarballs aren't restored yet. But the snapshot tarballs are found at: ftp://ftp.suse.com/pub/people/tiwai/snapshot/ Sorry for the late announce. thanks, Takashi

1 0

[alsa-devel] [PATCH v8 3/3] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 01 Nov '11

01 Nov '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for AUX1, AUX2 and INPGA (Moved here from DAPM patch) Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0ebcbd5..ff084a6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

3 8

[alsa-devel] [PATCH v3] ASoC: Add new Realtek ALC5632 CODEC driver.
by Leon Romanovsky 01 Nov '11

01 Nov '11

This driver implements basic functionality, using I²C for the control channel. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> Signed-off-by: Andrey Danin <danindrey(a)mail.ru> --- v3: * DAC is now handled by DAPM * Power down control correct handling. * Fixed speaker dapm routes. * Remove unused variables. * Sync stream names in DAPM and codec dai. * Use update bits instead write to reg in power depop function. * Rename Route to AB-D Amp Mux * Add snd controls to codec driver. Don't register it in probe. * Add dapm widgets and routes to codec driver. Don't register it. * Don't force power for spk apm, main i2s. It is described in DAPM. * Set symmetric_rates flag to prevent different rates for playback and capture. * Removed commented DEBUG definition. * Use SOC infrastructure sync cache mechanism. * Remove non possible IF switch. * Fix indentation issues * Convert alc5632 MICBIAS to a supply widget * Remove codec input parameter * Remove VERSION define * Remove unnecessary dev_kfree() and update copyright notice * Control speaker amplifier with DAPM. * Let machine driver choose master/slave mode for the codec. * Clean up. removed dev_dbg. * Use devm_kzalloc/devm_kzfree for memory allocation/deallocation. * Fixed stereo headphones playback. * Set up correct name for Stereo DAC volume item. v2: Free from checkpatch warnings v1: Initial code drop --- --- include/sound/alc5632.h | 33 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/alc5632.c | 1157 ++++++++++++++++++++ sound/soc/codecs/alc5632.h | 243 +++++ 6 files changed, 3869 insertions(+), 0 deletions(-) create mode 100644 include/sound/alc5632.h create mode 100644 sound/soc/codecs/alc5632.c create mode 100644 sound/soc/codecs/alc5632.h diff --git a/include/sound/alc5632.h b/include/sound/alc5632.h new file mode 100644 index 0000000..b994350 --- /dev/null +++ b/include/sound/alc5632.h @@ -0,0 +1,33 @@ +/* +* alc5632.h -- Platform data for ALC5632 +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.h by Arnaud Patard +* +* 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 _INCLUDE_SOUND_ALC5632_H +#define _INCLUDE_SOUND_ALC5632_H + +struct alc5632_platform_data { + /* configure : */ + /* Lineout/Speaker Amps Vmid ratio control */ + /* enable/disable adc/dac high pass filters */ + unsigned int add_ctrl; + /* configure : */ + /* output to enable when jack is low */ + /* output to enable when jack is high */ + /* jack detect (gpio/nc/jack detect [12] */ + unsigned int jack_det_ctrl; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..cd0aa46 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C + select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -169,6 +170,9 @@ config SND_SOC_AK4671 config SND_SOC_ALC5623 tristate +config SND_SOC_ALC5632 + tristate + config SND_SOC_CQ0093VC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..98bf1b6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o +snd-soc-alc5632-objs := alc5632.0 snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -113,6 +114,7 @@ obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c new file mode 100644 index 0000000..d4383f7 --- /dev/null +++ b/sound/soc/codecs/alc5632.c @@ -0,0 +1,1157 @@ +/* +* alc5632.c -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.c by Arnaud Patard +* +* 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/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/alc5632.h> + +#include "alc5632.h" + +/* + * ALC5632 register cache + */ +static const u16 alc5632_reg_defaults[] = { + 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ + 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ + 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ + 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ + 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ + 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ + 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ + 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ + 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ + 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ + 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ + 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ + 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ + 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ + 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ + 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ + 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +}; + +/* codec private data */ +struct alc5632_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; + unsigned int add_ctrl; + unsigned int jack_det_ctrl; +}; + +static int alc5632_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ALC5632_RESET: + case ALC5632_PWR_DOWN_CTRL_STATUS: + case ALC5632_GPIO_PIN_STATUS: + case ALC5632_OVER_CURR_STATUS: + case ALC5632_HID_CTRL_DATA: + case ALC5632_EQ_CTRL: + return 1; + + default: + break; + } + + return 0; +} + +static inline int alc5632_reset(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ALC5632_RESET, 0); + return snd_soc_read(codec, ALC5632_RESET); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5632 Controls + */ + +/* -34.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +/* -46.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +/* -16.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +/* 0db min scale, 6 db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); +/* 0db min scalem 0.75db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); + +static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { + /* left starts at bit 8, right at bit 0 */ + /* 31 steps (5 bit), -46.5db scale */ + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + /* bit 15 mutes left, bit 7 right */ + SOC_DOUBLE("Line Playback Switch", + ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5632_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Voice DAC Playback Volume", + ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE_TLV("Phone Capture Volume", + ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Stereo DAC Playback Volume", + ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), + SOC_DOUBLE("Stereo DAC Playback Switch", + ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1), +SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1), +SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1), +SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 9, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1), +SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 10, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1), +SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5632_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Mute"}; +static const char *alc5632_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5632_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5632_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5632_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5632_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static const struct snd_kcontrol_new alc5632_auxout_mux_controls = +SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5632_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static const struct snd_kcontrol_new alc5632_spkout_mux_controls = +SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5632_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = +SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5632_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = +SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5632_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = +SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); + +/* speaker amplifier */ +static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5632_amp_enum = + SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static const struct snd_kcontrol_new alc5632_amp_mux_controls = + SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); + + +static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5632_hp_mixer_controls[0], + ARRAY_SIZE(alc5632_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0, + &alc5632_hpr_mixer_controls[0], + ARRAY_SIZE(alc5632_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0, + &alc5632_hpl_mixer_controls[0], + ARRAY_SIZE(alc5632_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, + &alc5632_mono_mixer_controls[0], + ARRAY_SIZE(alc5632_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, + &alc5632_speaker_mixer_controls[0], + ARRAY_SIZE(alc5632_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, + &alc5632_captureL_mixer_controls[0], + ARRAY_SIZE(alc5632_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, + &alc5632_captureR_mixer_controls[0], + ARRAY_SIZE(alc5632_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0), +SND_SOC_DAPM_MIXER("DAC Right Channel", ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", ALC5632_PWR_MANAG_ADD1, 10, 0, + &alc5632_amp_mux_controls), + +SND_SOC_DAPM_OUTPUT("AUXOUT"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("PHONEP"), +SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + + +static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"Phone Mix", NULL, "Phone"}, + {"Phone Mix", NULL, "Phone ADMix"}, + {"AUXOUT", NULL, "Aux Out"}, + + /* DAC */ + {"DAC Right Channel", NULL, "I2S Mix"}, + {"DAC Left Channel", NULL, "I2S Mix"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + + {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, + {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, + + + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Aux Out", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Phone", NULL, "PHONEP"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"SpeakerOut N Mux", "RN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Right Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"Left Speaker", NULL, "AB-D Amp Mux"}, + {"Right Speaker", NULL, "AB-D Amp Mux"}, + + {"SPKOUT", NULL, "Left Speaker"}, + {"SPKOUT", NULL, "Right Speaker"}, + + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, + +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5632 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +/* FOUT = MCLK*(N+2)/((M+2)*(K+2)) + N: bit 15:8 (div 2 .. div 257) + K: bit 6:4 typical 2 + M: bit 3:0 (div 2 .. div 17) + + same as for 5623 - thanks! +*/ + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5632_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK) + return -ENODEV; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5632_DAI_CONTROL); + if (reg & ALC5632_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5632_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_BCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5632_PLL_FR_BCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_VBCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from voice clock */ + gbl_clk = ALC5632_PLL_FR_VBCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + + /* choose MCLK/BCLK/VBCLK */ + snd_soc_write(codec, ALC5632_GPCR2, gbl_clk); + /* choose PLL1 clock rate */ + snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div); + /* enable PLL1 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + ALC5632_PWR_ADD2_PLL1); + /* enable PLL2 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + ALC5632_PWR_ADD2_PLL2); + /* use PLL1 as main SYSCLK */ + snd_soc_update_bits(codec, ALC5632_GPCR1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {512*1, 0x3075}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5632->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5632->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5632_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5632_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5632_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5632_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5632_DAI_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5632_DAI_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); +} + +static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5632_DAI_CONTROL); + iface &= ~ALC5632_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5632_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5632_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5632_DAI_I2S_DL_24; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff); + + return 0; +} + +static int alc5632_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L \ + |ALC5632_MISC_HP_DEPOP_MUTE_R; + u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg); +} + +#define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF) + +#define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5632_ADD1_POWER_EN \ + (ALC5632_PWR_ADD1_DAC_REF \ + | ALC5632_PWR_ADD1_SOFTGEN_EN \ + | ALC5632_PWR_ADD1_HP_OUT_AMP \ + | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \ + | ALC5632_PWR_ADD1_MAIN_BIAS) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_SOFTGEN_EN, + ALC5632_PWR_ADD1_SOFTGEN_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD3, + ALC5632_ADD3_POWER_EN, + ALC5632_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + ALC5632_MISC_HP_DEPOP_MODE2_EN); + + msleep(500); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_ADD2_POWER_EN, + ALC5632_ADD2_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_ADD1_POWER_EN, + ALC5632_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5632_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_MAIN_BIAS); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_VREF); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, 0); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD3, 0); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5632_dai_ops = { + .hw_params = alc5632_pcm_hw_params, + .digital_mute = alc5632_mute, + .set_fmt = alc5632_set_dai_fmt, + .set_sysclk = alc5632_set_dai_sysclk, + .set_pll = alc5632_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5632_dai = { + .name = "alc5632-hifi", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + + .ops = &alc5632_dai_ops, + .symmetric_rates = 1, +}; + +static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5632_resume(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_cache_sync(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ + | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ + | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ + | ALC5632_ADC_REC_MONOMIX) + +#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ + | ALC5632_MIC_ROUTE_SPK \ + | ALC5632_MIC_ROUTE_MONOMIX) + +#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ + | ALC5632_PWR_DAC_STATUS \ + | ALC5632_PWR_AMIX_STATUS \ + | ALC5632_PWR_VREF_STATUS) + +#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ + / ALC5632_ADC_REC_GAIN_STEP) + +#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) + +#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) + +static int alc5632_probe(struct snd_soc_codec *codec) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5632_reset(codec); + + /* power on device */ + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (alc5632->add_ctrl) { + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, + alc5632->add_ctrl); + } + + /* "normal" mode: 0 @ 26 */ + /* set all PR0-7 mixers to 0 */ + snd_soc_update_bits(codec, ALC5632_PWR_DOWN_CTRL_STATUS, 0xEF00, 0); + + + /* power on VREF on all analog circuits 0x2000 @ 3C */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + 0, ALC5632_PWR_ADD2_VREF); + + switch (alc5632->id) { + case 0x5c: + snd_soc_add_controls(codec, alc5632_vol_snd_controls, + ARRAY_SIZE(alc5632_vol_snd_controls)); + break; + default: + return -EINVAL; + } + + return ret; +} + +/* power down chip */ +static int alc5632_remove(struct snd_soc_codec *codec) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5632 = { + .probe = alc5632_probe, + .remove = alc5632_remove, + .suspend = alc5632_suspend, + .resume = alc5632_resume, + .set_bias_level = alc5632_set_bias_level, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = alc5632_reg_defaults, + .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_register = alc5632_volatile_register, + .controls = alc5632_snd_controls, + .num_controls = ARRAY_SIZE(alc5632_snd_controls), + .dapm_widgets = alc5632_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(alc5632_dapm_widgets), + .dapm_routes = alc5632_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(alc5632_dapm_routes), +}; + +/* + * alc5632 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5632_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5632_platform_data *pdata; + struct alc5632_priv *alc5632; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_err(&client->dev, "got vid1: %x\n", vid1); + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_err(&client->dev, "got vid2: %x\n", vid2); + } + vid2 = (vid2 & 0xff); + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + alc5632 = devm_kzalloc(&client->dev, sizeof(struct alc5632_priv), GFP_KERNEL); + if (alc5632 == NULL) + return -ENOMEM; + + pdata = client->dev.platform_data; + if (pdata) { + alc5632->add_ctrl = pdata->add_ctrl; + alc5632->jack_det_ctrl = pdata->jack_det_ctrl; + } + + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + return -EINVAL; + } + + i2c_set_clientdata(client, alc5632); + alc5632->control_data = client; + alc5632->control_type = SND_SOC_I2C; + mutex_init(&alc5632->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5632, &alc5632_dai, 1); + if (ret != 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + + return ret; +} + +static int alc5632_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id alc5632_i2c_table[] = { + {"alc5632", 0x5c}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5632_i2c_driver = { + .driver = { + .name = "alc5632", + .owner = THIS_MODULE, + }, + .probe = alc5632_i2c_probe, + .remove = __devexit_p(alc5632_i2c_remove), + .id_table = alc5632_i2c_table, +}; + +static int __init alc5632_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5632_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5632_modinit); + +static void __exit alc5632_modexit(void) +{ + i2c_del_driver(&alc5632_i2c_driver); +} +module_exit(alc5632_modexit); + +MODULE_DESCRIPTION("ASoC ALC5632 driver"); +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h new file mode 100644 index 0000000..9747016 --- /dev/null +++ b/sound/soc/codecs/alc5632.h @@ -0,0 +1,243 @@ +/* +* alc5632.h -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.h by Arnaud Patard +* +* 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 _ALC5632_H +#define _ALC5632_H + +#define ALC5632_RESET 0x00 +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5632_SPK_OUT_VOL 0x02 /* spe out vol */ +#define ALC5632_SPK_OUT_VOL_STEP 1.5 +#define ALC5632_HP_OUT_VOL 0x04 /* hp out vol */ +#define ALC5632_AUX_OUT_VOL 0x06 /* aux out vol */ +#define ALC5632_PHONE_IN_VOL 0x08 /* phone in vol */ +#define ALC5632_LINE_IN_VOL 0x0A /* line in vol */ +#define ALC5632_STEREO_DAC_IN_VOL 0x0C /* stereo dac in vol */ +#define ALC5632_MIC_VOL 0x0E /* mic in vol */ +/* stero dac/mic routing */ +#define ALC5632_MIC_ROUTING_CTRL 0x10 +#define ALC5632_MIC_ROUTE_MONOMIX (1 << 0) +#define ALC5632_MIC_ROUTE_SPK (1 << 1) +#define ALC5632_MIC_ROUTE_HP (1 << 2) + +#define ALC5632_ADC_REC_GAIN 0x12 /* rec gain */ +#define ALC5632_ADC_REC_GAIN_RANGE 0x1F1F +#define ALC5632_ADC_REC_GAIN_BASE (-16.5) +#define ALC5632_ADC_REC_GAIN_STEP 1.5 + +#define ALC5632_ADC_REC_MIXER 0x14 /* mixer control */ +#define ALC5632_ADC_REC_MIC1 (1 << 6) +#define ALC5632_ADC_REC_MIC2 (1 << 5) +#define ALC5632_ADC_REC_LINE_IN (1 << 4) +#define ALC5632_ADC_REC_AUX (1 << 3) +#define ALC5632_ADC_REC_HP (1 << 2) +#define ALC5632_ADC_REC_SPK (1 << 1) +#define ALC5632_ADC_REC_MONOMIX (1 << 0) + +#define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */ +/* ALC5632_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */ +#define ALC5632_OUTPUT_MIXER_RP (1 << 14) +#define ALC5632_OUTPUT_MIXER_WEEK (1 << 12) +#define ALC5632_OUTPUT_MIXER_HP (1 << 10) +#define ALC5632_OUTPUT_MIXER_AUX_SPK (2 << 6) +#define ALC5632_OUTPUT_MIXER_AUX_HP_LR (1 << 6) +#define ALC5632_OUTPUT_MIXER_HP_R (1 << 8) +#define ALC5632_OUTPUT_MIXER_HP_L (1 << 9) + +#define ALC5632_MIC_CTRL 0x22 /* mic phone ctrl */ +#define ALC5632_MIC_BOOST_BYPASS 0 +#define ALC5632_MIC_BOOST_20DB 1 +#define ALC5632_MIC_BOOST_30DB 2 +#define ALC5632_MIC_BOOST_40DB 3 + +#define ALC5632_DIGI_BOOST_CTRL 0x24 /* digi mic / bost ctl */ +#define ALC5632_MIC_BOOST_RANGE 7 +#define ALC5632_MIC_BOOST_STEP 6 +#define ALC5632_PWR_DOWN_CTRL_STATUS 0x26 +#define ALC5632_PWR_VREF_STATUS (1 << 3) +#define ALC5632_PWR_AMIX_STATUS (1 << 2) +#define ALC5632_PWR_DAC_STATUS (1 << 1) +#define ALC5632_PWR_ADC_STATUS (1 << 0) +/* stereo/voice DAC / stereo adc func ctrl */ +#define ALC5632_DAC_FUNC_SELECT 0x2E + +/* Main serial data port ctrl (i2s) */ +#define ALC5632_DAI_CONTROL 0x34 + +#define ALC5632_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5632_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5632_DAI_SADLRCK_MODE (1 << 14) +/* 0:voice, 1:main */ +#define ALC5632_DAI_MAIN_I2S_SYSCLK_SEL (1 << 8) +#define ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_MAIN_I2S_LRCK_INV (1 << 6) +#define ALC5632_DAI_I2S_DL_MASK (3 << 2) +#define ALC5632_DAI_I2S_DL_8 (3 << 2) +#define ALC5632_DAI_I2S_DL_24 (2 << 2) +#define ALC5632_DAI_I2S_DL_20 (1 << 2) +#define ALC5632_DAI_I2S_DL_16 (0 << 2) +#define ALC5632_DAI_I2S_DF_MASK (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_B (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_A (2 << 0) +#define ALC5632_DAI_I2S_DF_LEFT (1 << 0) +#define ALC5632_DAI_I2S_DF_I2S (0 << 0) +/* extend serial data port control (VoDAC_i2c/pcm) */ +#define ALC5632_DAI_CONTROL2 0x36 +/* 0:gpio func, 1:voice pcm */ +#define ALC5632_DAI_VOICE_PCM_ENABLE (1 << 15) +/* 0:master, 1:slave */ +#define ALC5632_DAI_VOICE_MODE_SEL (1 << 14) +/* 0:disable, 1:enable */ +#define ALC5632_DAI_HPF_CLK_CTRL (1 << 13) +/* 0:main, 1:voice */ +#define ALC5632_DAI_VOICE_I2S_SYSCLK_SEL (1 << 8) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_VBCLK_SYSCLK_SEL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_I2S_LR_INV (1 << 6) +#define ALC5632_DAI_VOICE_DL_MASK (3 << 2) +#define ALC5632_DAI_VOICE_DL_16 (0 << 2) +#define ALC5632_DAI_VOICE_DL_20 (1 << 2) +#define ALC5632_DAI_VOICE_DL_24 (2 << 2) +#define ALC5632_DAI_VOICE_DL_8 (3 << 2) +#define ALC5632_DAI_VOICE_DF_MASK (3 << 0) +#define ALC5632_DAI_VOICE_DF_I2S (0 << 0) +#define ALC5632_DAI_VOICE_DF_LEFT (1 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_A (2 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_B (3 << 0) + +#define ALC5632_PWR_MANAG_ADD1 0x3A +#define ALC5632_PWR_ADD1_DAC_L_EN (1 << 15) +#define ALC5632_PWR_ADD1_DAC_R_EN (1 << 14) +#define ALC5632_PWR_ADD1_ZERO_CROSS (1 << 13) +#define ALC5632_PWR_ADD1_MAIN_I2S_EN (1 << 11) +#define ALC5632_PWR_ADD1_SPK_AMP_EN (1 << 10) +#define ALC5632_PWR_ADD1_HP_OUT_AMP (1 << 9) +#define ALC5632_PWR_ADD1_HP_OUT_ENH_AMP (1 << 8) +#define ALC5632_PWR_ADD1_VOICE_DAC_MIX (1 << 7) +#define ALC5632_PWR_ADD1_SOFTGEN_EN (1 << 6) +#define ALC5632_PWR_ADD1_MIC1_SHORT_CURR (1 << 5) +#define ALC5632_PWR_ADD1_MIC2_SHORT_CURR (1 << 4) +#define ALC5632_PWR_ADD1_MIC1_EN (1 << 3) +#define ALC5632_PWR_ADD1_MIC2_EN (1 << 2) +#define ALC5632_PWR_ADD1_MAIN_BIAS (1 << 1) +#define ALC5632_PWR_ADD1_DAC_REF (1 << 0) + +#define ALC5632_PWR_MANAG_ADD2 0x3C +#define ALC5632_PWR_ADD2_PLL1 (1 << 15) +#define ALC5632_PWR_ADD2_PLL2 (1 << 14) +#define ALC5632_PWR_ADD2_VREF (1 << 13) +#define ALC5632_PWR_ADD2_OVT_DET (1 << 12) +#define ALC5632_PWR_ADD2_VOICE_DAC (1 << 10) +#define ALC5632_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5632_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5632_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5632_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5632_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5632_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5632_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5632_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5632_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5632_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5632_PWR_MANAG_ADD3 0x3E +#define ALC5632_PWR_ADD3_AUXOUT_VOL (1 << 14) +#define ALC5632_PWR_ADD3_SPK_L_OUT (1 << 13) +#define ALC5632_PWR_ADD3_SPK_R_OUT (1 << 12) +#define ALC5632_PWR_ADD3_HP_L_OUT_VOL (1 << 11) +#define ALC5632_PWR_ADD3_HP_R_OUT_VOL (1 << 10) +#define ALC5632_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5632_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5632_PWR_ADD3_AUXIN_VOL (1 << 5) +#define ALC5632_PWR_ADD3_AUXIN_MIX (1 << 4) +#define ALC5632_PWR_ADD3_MIC1_VOL (1 << 3) +#define ALC5632_PWR_ADD3_MIC2_VOL (1 << 2) +#define ALC5632_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5632_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5632_GPCR1 0x40 +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1 (1 << 15) +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_MCLK (0 << 15) +#define ALC5632_GPCR1_DAC_HI_FLT_EN (1 << 10) +#define ALC5632_GPCR1_SPK_AMP_CTRL (7 << 1) +#define ALC5632_GPCR1_VDD_100 (5 << 1) +#define ALC5632_GPCR1_VDD_125 (4 << 1) +#define ALC5632_GPCR1_VDD_150 (3 << 1) +#define ALC5632_GPCR1_VDD_175 (2 << 1) +#define ALC5632_GPCR1_VDD_200 (1 << 1) +#define ALC5632_GPCR1_VDD_225 (0 << 1) + +#define ALC5632_GPCR2 0x42 +#define ALC5632_GPCR2_PLL1_SOUR_SEL (3 << 12) +#define ALC5632_PLL_FR_MCLK (0 << 12) +#define ALC5632_PLL_FR_BCLK (2 << 12) +#define ALC5632_PLL_FR_VBCLK (3 << 12) +#define ALC5632_GPCR2_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5632_PLL1_CTRL 0x44 +#define ALC5632_PLL1_CTRL_N_VAL(n) (((n) & 0x0f) << 8) +#define ALC5632_PLL1_M_BYPASS (1 << 7) +#define ALC5632_PLL1_CTRL_K_VAL(k) (((k) & 0x07) << 4) +#define ALC5632_PLL1_CTRL_M_VAL(m) (((m) & 0x0f) << 0) + +#define ALC5632_PLL2_CTRL 0x46 +#define ALC5632_PLL2_EN (1 << 15) +#define ALC5632_PLL2_RATIO (0 << 15) + +#define ALC5632_GPIO_PIN_CONFIG 0x4C +#define ALC5632_GPIO_PIN_POLARITY 0x4E +#define ALC5632_GPIO_PIN_STICKY 0x50 +#define ALC5632_GPIO_PIN_WAKEUP 0x52 +#define ALC5632_GPIO_PIN_STATUS 0x54 +#define ALC5632_GPIO_PIN_SHARING 0x56 +#define ALC5632_OVER_CURR_STATUS 0x58 +#define ALC5632_SOFTVOL_CTRL 0x5A +#define ALC5632_GPIO_OUPUT_PIN_CTRL 0x5C + +#define ALC5632_MISC_CTRL 0x5E +#define ALC5632_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5632_MISC_AVC_TRGT_SEL (3 << 12) +#define ALC5632_MISC_AVC_TRGT_RIGHT (1 << 12) +#define ALC5632_MISC_AVC_TRGT_LEFT (2 << 12) +#define ALC5632_MISC_AVC_TRGT_BOTH (3 << 12) +#define ALC5632_MISC_HP_DEPOP_MODE1_EN (1 << 9) +#define ALC5632_MISC_HP_DEPOP_MODE2_EN (1 << 8) +#define ALC5632_MISC_HP_DEPOP_MUTE_L (1 << 7) +#define ALC5632_MISC_HP_DEPOP_MUTE_R (1 << 6) +#define ALC5632_MISC_HP_DEPOP_MUTE (1 << 5) +#define ALC5632_MISC_GPIO_WAKEUP_CTRL (1 << 1) +#define ALC5632_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5632_DAC_CLK_CTRL1 0x60 +#define ALC5632_DAC_CLK_CTRL2 0x62 +#define ALC5632_DAC_CLK_CTRL2_DIV1_2 (1 << 0) +#define ALC5632_VOICE_DAC_PCM_CLK_CTRL1 0x64 +#define ALC5632_PSEUDO_SPATIAL_CTRL 0x68 +#define ALC5632_HID_CTRL_INDEX 0x6A +#define ALC5632_HID_CTRL_DATA 0x6C +#define ALC5632_EQ_CTRL 0x6E + +/* undocumented */ +#define ALC5632_VENDOR_ID1 0x7C +#define ALC5632_VENDOR_ID2 0x7E + +#endif -- 1.7.3.4

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Re: [alsa-devel] [PATCH] hda_hwdep: Fix possible buffer overflow
by Takashi Iwai 01 Nov '11

01 Nov '11

At Wed, 26 Oct 2011 15:15:24 +0200, Alexander Stein wrote: > > Hello, > > On Wednesday 26 October 2011 14:58:43 Takashi Iwai wrote: > > At Wed, 26 Oct 2011 09:48:12 +0200, > > > > Alexander Stein wrote: > > > If a line in the firmware file is larger than the given buffer size (and > > > so the firmware file size), size is set to a value larger than the actual > > > buffer size. This results in an overflow in the buffer passed. > > > Fix this by copying only up to 127 chars per line. > > > > Actually this check should have been > > > > if (size > fw->size) > > size = fw->size; > > > > Otherwise it doesn't make sense. > > If the change is OK, could you resend the patch with it? > > IMO this check isn't even needed. This case should be catched by this check > > for (len = 0; len < fw->size; len++) { > > already. > Opinions? Right, it's superfluous. Let's get rid of it. Takashi

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[alsa-devel] [PATCH] ASoC: Fix return value of wm5100_gpio_direction_out()
by Mark Brown 31 Oct '11

31 Oct '11

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 3ab725a..cf66209 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2366,13 +2366,17 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip, { struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); struct snd_soc_codec *codec = wm5100->codec; - int val; + int val, ret; val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); - return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, - WM5100_GP1_FN_MASK | WM5100_GP1_DIR | - WM5100_GP1_LVL, val); + ret = snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset, + WM5100_GP1_FN_MASK | WM5100_GP1_DIR | + WM5100_GP1_LVL, val); + if (ret < 0) + return ret; + else + return 0; } static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) -- 1.7.7.1

1 0

[alsa-devel] No line out on latest Mac Mini
by John Frankish 31 Oct '11

31 Oct '11

I posted this to the alsa-user list a couple of weeks ago but did not get an answer I've been trying to get analog line out to work to headphones or powered speakers with the latest Mac Mini, but so far the best I've managed to achieve is to get the internal speaker to work. Line out works with the Mac OS. I've also tried the HDA analyser mentioned here: http://www.alsa-project.org/main/index.php/HDA_Analyzer but no matter what I do, there is no way to get sound from the line out jack to headphones or powered speakers. I'm guessing I need one or more obscure options to load the module snd-hda-intel, but nothing I've tried so far has worked. The kernel documentation does not mention the mac mini 5,3 the Intel Corporation 6 Series/C200 Series Chipset nor the combination of Cirrus Logic CS4206 and Intel CougarPoint HDMI - does anybody know where I might find more details? Details below: ---------- DMI: Apple Inc. Macmini5,3/Mac-7BA5B2794B2CDB12, BIOS MM51.88Z.0075.B00.1106271442 06/27/2011 Alsa modules from linux-3.0.3 Alsa-1.0.23 $ lspci -knn 00:1b.0 Audio device [0403]: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller [8086:1c20] (rev 05) Subsystem: Intel Corporation Device [8086:7270] Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: Cirrus Analog [Cirrus Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: Cirrus Digital [Cirrus Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 $ head -n 1 /proc/asound/card0/codec* ==> /proc/asound/card0/codec#0 <== Codec: Cirrus Logic CS4206 ==> /proc/asound/card0/codec#3 <== Codec: Intel CougarPoint HDMI

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[alsa-devel] [PATCH] ASoC: WM8904: Set `invert' bit for Capture Switch
by Hong Xu 31 Oct '11

31 Oct '11

Set `invert' bit for Capture Switch. Otherwise analogue is muted when Capture Switch is ON. Signed-off-by: Hong Xu <hong.xu(a)atmel.com> --- sound/soc/codecs/wm8904.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b085575..cbba0b1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -868,7 +868,7 @@ SOC_ENUM("Right Capture Mode", rin_mode), SOC_DOUBLE_R("Capture Volume", WM8904_ANALOGUE_LEFT_INPUT_0, WM8904_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, - WM8904_ANALOGUE_RIGHT_INPUT_0, 7, 1, 0), + WM8904_ANALOGUE_RIGHT_INPUT_0, 7, 1, 1), SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("High Pass Filter Mode", hpf_mode), -- 1.7.3.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8711: Fix wrong mask for setting input audio data bit length select
by Axel Lin 31 Oct '11

31 Oct '11

The Input Audio Data Bit Length Select is controlled by BIT[3:2] of WM8711_IFACE(07h) register. Current code incorrectly masks BIT[1:0] which is for Audio Data Format Select. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8711.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 8d0347c..8ba49d8 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -151,7 +151,7 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec); - u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc; + u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfff3; int i = get_coeff(wm8711->sysclk, params_rate(params)); u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | coeff_div[i].usb; -- 1.7.5.4

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[alsa-devel] [PATCH 1/3] ALSA: hdspm - Fix MADI channel format in the status ioctl
by Adrian Knoth 31 Oct '11

31 Oct '11

SNDRV_HDSPM_IOCTL_GET_STATUS is supposed to query the current card status, so we have to return what we receive on the MADI wire (RX), not what we transmit (TX) to others. The latter is a config item to be queried via SNDRV_HDSPM_IOCTL_GET_CONFIG. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6e2f7ef..60a0b7d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6253,7 +6253,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, status.card_specific.madi.madi_input = (statusregister & HDSPM_AB_int) ? 1 : 0; status.card_specific.madi.channel_format = - (statusregister & HDSPM_TX_64ch) ? 1 : 0; + (statusregister & HDSPM_RX_64ch) ? 1 : 0; /* TODO: Mac driver sets it when f_s>48kHz */ status.card_specific.madi.frame_format = 0; -- 1.7.7.1

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Re: [alsa-devel] [PATCH 1/1] intel8x0: [v3] Improve performance in virtual environment
by Takashi Iwai 31 Oct '11

31 Oct '11

At Wed, 26 Oct 2011 19:11:01 +0400, Konstantin Ozerkov wrote: > > v3: detection code is x86 and KVM specific, hide it under ifdef > v2: add detection for virtual environments (KVM and Parallels) > > This patch is intended to improve performance in virtualized environments > like Parallels Desktop or KVM/VirtualBox/QEMU (virtual ICH/AC97 audio). > > I/O access is very time-expensive operation in virtual world: VCPU > can be rescheduled and in the worst case we get more than 10ms delay on > each I/O access. > > In the virtual environment loop exit rule > (old_civ == current_civ && old_picb == current_picb) is never satisfied, > because old_picb is never the same as current_picb due to delay inspired > by reading current_civ. As a result loop ended by timeout and we get 10x > more I/O operations. > > Experimental data from Prallels Desktop 7, RHEL6 guest (I/O ops per > second): > > Original code: > In Port Counter Callback > f014 41550 fffff00000179d00 ac97_bm_read_civ+0x000 > f018 41387 fffff0000017a580 ac97_bm_read_picb+0x000 > > With patch: > In Port Counter Callback > f014 4090 fffff00000179d00 ac97_bm_read_civ+0x000 > f018 1964 fffff0000017a580 ac97_bm_read_picb+0x000 > > Signed-off-by: Konstantin Ozerkov <kozerkov(a)parallels.com> > Signed-off-by: Denis V. Lunev <den(a)openvz.org> Thanks, applied now. Takashi > > diff --git a/pci/intel8x0.c b/pci/intel8x0.c > index 6a5b387..6dc302c 100644 > --- a/pci/intel8x0.c > +++ b/pci/intel8x0.c > @@ -42,6 +42,12 @@ > #include <asm/pgtable.h> > #include <asm/cacheflush.h> > > +#ifdef CONFIG_KVM_GUEST > +#include <asm/kvm_para.h> > +#else > +#define kvm_para_available() (0) > +#endif > + > MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); > MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); > MODULE_LICENSE("GPL"); > @@ -77,6 +83,7 @@ static int buggy_semaphore; > static int buggy_irq = -1; /* auto-check */ > static int xbox; > static int spdif_aclink = -1; > +static int inside_vm = -1; > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for Intel i8x0 soundcard."); > @@ -94,6 +101,8 @@ module_param(xbox, bool, 0444); > MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection."); > module_param(spdif_aclink, int, 0444); > MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link."); > +module_param(inside_vm, bool, 0444); > +MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization."); > > /* just for backward compatibility */ > static int enable; > @@ -400,6 +409,7 @@ struct intel8x0 { > unsigned buggy_irq: 1; /* workaround for buggy mobos */ > unsigned xbox: 1; /* workaround for Xbox AC'97 detection */ > unsigned buggy_semaphore: 1; /* workaround for buggy codec semaphore */ > + unsigned inside_vm: 1; /* enable VM optimization */ > > int spdif_idx; /* SPDIF BAR index; *_SPBAR or -1 if use PCMOUT */ > unsigned int sdm_saved; /* SDM reg value */ > @@ -1065,8 +1075,11 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs > udelay(10); > continue; > } > - if (civ == igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV) && > - ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb)) > + if (civ != igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV)) > + continue; > + if (chip->inside_vm) > + break; > + if (ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb)) > break; > } while (timeout--); > ptr = ichdev->last_pos; > @@ -2984,6 +2997,10 @@ static int __devinit snd_intel8x0_create(struct snd_card *card, > if (xbox) > chip->xbox = 1; > > + chip->inside_vm = inside_vm; > + if (inside_vm) > + printk(KERN_INFO "intel8x0: enable KVM optimization\n"); > + > if (pci->vendor == PCI_VENDOR_ID_INTEL && > pci->device == PCI_DEVICE_ID_INTEL_440MX) > chip->fix_nocache = 1; /* enable workaround */ > @@ -3226,6 +3243,14 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci, > buggy_irq = 0; > } > > + if (inside_vm < 0) { > + /* detect KVM and Parallels virtual environments */ > + inside_vm = kvm_para_available(); > +#if defined(__i386__) || defined(__x86_64__) > + inside_vm = inside_vm || boot_cpu_has(X86_FEATURE_HYPERVISOR); > +#endif > + } > + > if ((err = snd_intel8x0_create(card, pci, pci_id->driver_data, > &chip)) < 0) { > snd_card_free(card); > -- > 1.7.7 >

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[alsa-devel] [patch v2] [ALSA] hwdep: silence integer overflow warning
by Dan Carpenter 31 Oct '11

31 Oct '11

Smatch complains that if device is INT_MAX then device + 1 can overflow. It just means we would have an annoying loop while we check all the devices from -2147483648 to SNDRV_MINOR_HWDEPS. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- v2: reshuffled it a bit per Walter Harms's suggestion. diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index c7ceb28..75ea16f 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -273,7 +273,14 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; mutex_lock(&register_mutex); - device = device < 0 ? 0 : device + 1; + + if (device < 0) + device = 0; + else if (device < SNDRV_MINOR_HWDEPS) + device++; + else + device = SNDRV_MINOR_HWDEPS; + while (device < SNDRV_MINOR_HWDEPS) { if (snd_hwdep_search(card, device)) break;

2 1

Re: [alsa-devel] [patch] [ALSA] hwdep: silence integer overflow warning
by Dan Carpenter 28 Oct '11

28 Oct '11

On Thu, Oct 20, 2011 at 09:39:23AM +0200, walter harms wrote: > I am not sure how SNDRV_MINOR_HWDEPS is used further Either device = SNDRV_MINOR_HWDEPS or device = SNDRV_MINOR_HWDEPS + 1 are fine (equivalent) but it's a fair point. Your version is nicer. I'll redo the patch. regards, dan carpenter

2 2

[alsa-devel] Aureal issues on resume
by Clemens Ladisch 27 Oct '11

27 Oct '11

(added recipients) -------- Original Message -------- Subject: Aureal issues on resume Date: Fri, 02 Sep 2011 13:41:42 +0200 From: Anders Eriksson <aeriksson(a)fastmail.fm> To: linux-kernel(a)vger.kernel.org, linux-sound(a)vger.kernel.org Hi, upon resuming from hibernation, I see this in my log: Sep 2 11:15:20 tippex kernel: [72436.287020] vortex_adb_addroutes: unending route! 0x10 Sep 2 11:15:20 tippex kernel: [72436.287127] vortex_adb_addroutes: unending route! 0x838 Sep 2 11:15:20 tippex kernel: [72436.287277] vortex_adb_addroutes: unending route! 0x11 Sep 2 11:15:20 tippex kernel: [72436.287418] vortex_adb_addroutes: unending route! 0x8b9 Sep 2 11:15:20 tippex kernel: [72436.292037] Vortex: vortex_fifo_setadbctrl fail Sep 2 11:15:55 tippex kernel: [72471.403490] vortex_adb_delroutes: route not found! 0x10 Sep 2 11:15:55 tippex kernel: [72471.403596] vortex_adb_delroutes: route not found! 0x838 Sep 2 11:15:55 tippex kernel: [72471.403710] vortex_adb_delroutes: route not found! 0x11 Sep 2 11:15:55 tippex kernel: [72471.403811] vortex_adb_delroutes: route not found! 0x8b9 And the sound is no more. I need to reboot to get it back. If memory serves, it's been there since 2.6.30 or so. Any ideas what can be triggering it? lspci says it's an 00:04.0 Multimedia audio controller: Aureal Semiconductor Vortex 1 (rev 02) Subsystem: Diamond Multimedia Systems Sonic Impact A3D Flags: bus master, medium devsel, latency 64, IRQ 11 Memory at 50000000 (32-bit, non-prefetchable) [size=128K] I/O ports at 10b0 [size=8] I/O ports at 10b8 [size=8] Capabilities: [dc] Power Management version 1 Kernel driver in use: snd_au8820 Kernel modules: snd-au8820 -Anders

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[alsa-devel] Help about LineIn capture on WM8904
by gnuser 27 Oct '11

27 Oct '11

Hi Mark and all, I'm working on an ASoC machine driver to include the support for WM8904. The ARM SoC is AT91SAM9G45 from Atmel. In Atmel's original reference board, Atmel uses WM8731 which works quite well. For my case, I replaced it with WM8904. Both Atmel's ref board and mine use I2S to talk with audio codec. So far, the audio playback is OK. But audio capture can not work well. The *hardware connection* in my case is, external 32768Hz crystal --> MCLK (Use FLL to get proper sysclk) Line In Jack --> IN2L Line In Jack --> IN2R MiC --> IN1L HeadPhone --> HPOUTL HeadPhone --> HPOUTR NC Pins: LINEOUTL, LINEOUTR, LINEOUTFB, IN1R Codec is in master mode, ARM Soc is in slave mode. My questions are, 1) Since WM8731 works fine (both playback and capture) on Atmel's ref board, and playback is also OK for my case, can we get the conclusion that the PCM driver and DAI driver are OK? Does the clock configuration for my case also look OK? 2) In my machine driver, I added the following DAPM components, static const struct snd_soc_dapm_widget mini_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_LINE("LineIn Jack", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* headphone jack connected to HPOUT */ {"Headphone Jack", NULL, "HPOUTL"}, {"Headphone Jack", NULL, "HPOUTR"}, {"MICBIAS", NULL, "Mic Jack"}, {"IN1L", NULL, "MICBIAS"}, {"IN2R", NULL, "LineIn Jack"}, {"IN2L", NULL, "LineIn Jack"}, }; - and - then, { snd_soc_dapm_nc_pin(dapm, "IN1R"); snd_soc_dapm_nc_pin(dapm, "IN3L"); snd_soc_dapm_nc_pin(dapm, "IN3R"); snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); snd_soc_dapm_new_controls(dapm, mini_dapm_widgets, ARRAY_SIZE(mini_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "LineIn Jack"); snd_soc_dapm_sync(dapm); } Does the above code look fine? 3) I used "arecord -d 8 -f S16_LE -c2 -r8000 8000.wav" to capture audio. But the captured data are something looks like 84 FF 73 FF 83 FF 70 FF 85 FF 72 FF 84 FF 71 FF 83 FF 72 FF 84 FF 72 FF 87 FF 71 FF 83 FF 71 FF (RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 8000 Hz) The captured file can not be played well (no sound actually) Both arecord, PCM driver, DAI driver and snd core driver are quiet without any warning or error. I tried to use alsamixer to configure the audo path. But I'm new to ALSA and not sure about the configuration. I just paste the amixer here, could you please point me where I've made mistakes? A piece of amixer scripts will greatly help me. :) Thanks in advance. Simple mixer control 'Headphone',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 63 Front Left: 46 [73%] [-11.00dB] Playback [on] Front Right: 45 [71%] [-12.00dB] Playback [on] Simple mixer control 'Headphone ZC',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Line Output',0 Capabilities: volume pswitch penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 63 Front Left: 57 [90%] [0.00dB] Playback [on] Front Right: 57 [90%] [0.00dB] Playback [on] Simple mixer control 'Line Output ZC',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 5 [16%] [on] Front Right: Capture 5 [16%] [on] Simple mixer control 'ADC 128x OSR',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'AIFOUTL Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Left' Simple mixer control 'AIFOUTR Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Right' Simple mixer control 'DAC Deemphasis',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'DAC OSRx2',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'DACL Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Left' Simple mixer control 'DACR Mux',0 Capabilities: enum Items: 'Left' 'Right' Item0: 'Right' Simple mixer control 'DRC',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'DRC Path',0 Capabilities: enum Items: 'ADC' 'DAC' Item0: 'ADC' Simple mixer control 'Digital',0 Capabilities: pvolume cvolume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 96 Capture 0 - 119 Front Left: Playback 96 [100%] [0.00dB] Capture 96 [81%] [0.00dB] Front Right: Playback 96 [100%] [0.00dB] Capture 96 [81%] [0.00dB] Simple mixer control 'Digital Playback Boost',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 0 [0%] [0.00dB] Simple mixer control 'Digital Sidetone',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 15 Front Left: 0 [0%] [-36.00dB] Front Right: 0 [0%] [-36.00dB] Simple mixer control 'EQ',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'EQ1',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ2',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ3',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ4',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'EQ5',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 24 Mono: 12 [50%] [0.00dB] Simple mixer control 'HPL Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'HPR Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'High Pass Filter',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'High Pass Filter Mode',0 Capabilities: enum Items: 'Hi-fi' 'Voice 1' 'Voice 2' 'Voice 3' Item0: 'Hi-fi' Simple mixer control 'LINEL Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'LINER Mux',0 Capabilities: enum Items: 'DAC' 'Bypass' Item0: 'DAC' Simple mixer control 'Left Capture Inverting Mux',0 Capabilities: enum Items: 'IN1L' 'IN2L' 'IN3L' Item0: 'IN1L' Simple mixer control 'Left Capture Mux',0 Capabilities: enum Items: 'IN1L' 'IN2L' 'IN3L' Item0: 'IN2L' Simple mixer control 'Left Caputure Mode',0 Capabilities: enum Items: 'Single-Ended' 'Differential Line' 'Differential Mic' Item0: 'Single-Ended' Simple mixer control 'Left Sidetone',0 Capabilities: enum Items: 'None' 'Left' 'Right' Item0: 'None' Simple mixer control 'Right Capture Inverting Mux',0 Capabilities: enum Items: 'IN1R' 'IN2R' 'IN3R' Item0: 'IN1R' Simple mixer control 'Right Capture Mode',0 Capabilities: enum Items: 'Single-Ended' 'Differential Line' 'Differential Mic' Item0: 'Single-Ended' Simple mixer control 'Right Capture Mux',0 Capabilities: enum Items: 'IN1R' 'IN2R' 'IN3R' Item0: 'IN2R' Simple mixer control 'Right Sidetone',0 Capabilities: enum Items: 'None' 'Left' 'Right' Item0: 'None'

2 1

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic23: Clear TLV320AIC23_MS_MASTER bit for slave mode
by Axel Lin 27 Oct '11

27 Oct '11

According to the datasheet: Digital Audio Interface Format (07h) register: BIT6: Master/slave mode 0: Slave 1: Master Current code sets TLV320AIC23_MS_MASTER bit for master mode, but does not clear it for slave mode. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic23.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index ab27dbc..336de8f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -430,6 +430,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, iface_reg |= TLV320AIC23_MS_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + iface_reg &= ~TLV320AIC23_MS_MASTER; break; default: return -EINVAL; -- 1.7.5.4

4 4

[alsa-devel] [PATCH] ASoC: Remove needless unlikely()
by Mark Brown 27 Oct '11

27 Oct '11

There's no point in adding unlikely() annotations outside of hot paths and on systems using these features the annotation will always be wrong (as opposed to being something that only comes up once in a while) so the annotation may even be harmful. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 52a7259..49aa71e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,8 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time)) { + if (codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC:core:add flag ignore_pmdown_time in snd_soc_dai_link
by ramesh.babu＠linux.intel.com 27 Oct '11

27 Oct '11

Resending the patch by CCing Peter With this flag, each dai_link in machine driver can choose to ignore pmdown_time during DAPM shut down sequence. If the ignore_pmdown_time is set, the DAPM for corresponding DAI will be executed immediately. Signed-off-by: Ramesh Babu K V <ramesh.babu(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 3 ++- 2 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 11cfb59..877fcc1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -718,6 +718,9 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; + /* pmdown_time is ignored at stop */ + unsigned int ignore_pmdown_time:1; + /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee15337..52a7259 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,7 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time)) { + if (unlikely(codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2] ASoC: wm8940: Fix mask for setting BCLKDIV
by Axel Lin 27 Oct '11

27 Oct '11

According to the datasheet: BCLK is controlled by BIT[4:2] of WM8940_CLOCK(06h) register. Signed-off-by: Chris Paulson-Ellis <chris(a)edesix.com> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8940.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index dc5cb31..de9ec9b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -621,7 +621,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8940_BCLKDIV: - reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3; + reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFE3; ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2)); break; case WM8940_MCLKDIV: -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Use snd_soc_update_bits for read-modify-write
by Axel Lin 27 Oct '11

27 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 41 +++++++++++++-------------------------- 1 files changed, 14 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7a49390..a77f6ea 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -833,7 +833,6 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; u16 d, pll_d = 1; - u8 reg; int clk; /* select data word length */ @@ -869,14 +868,13 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT); snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV); /* disable PLL if it is bypassed */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg & ~PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLL_ENABLE, 0); } else { snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV); /* enable PLL when it is used */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, reg | PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); } /* Route Left DAC to left channel input and @@ -1155,7 +1153,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - u8 reg; switch (level) { case SND_SOC_BIAS_ON: @@ -1164,9 +1161,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY && aic3x->master) { /* enable pll */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - reg | PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, PLL_ENABLE); } break; case SND_SOC_BIAS_STANDBY: @@ -1175,9 +1171,8 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE && aic3x->master) { /* disable pll */ - reg = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - reg & ~PLL_ENABLE); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, + PLL_ENABLE, 0); } break; case SND_SOC_BIAS_OFF: @@ -1294,7 +1289,6 @@ static int aic3x_resume(struct snd_soc_codec *codec) static int aic3x_init(struct snd_soc_codec *codec) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - int reg; snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); @@ -1316,20 +1310,13 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* unmute all outputs */ - reg = snd_soc_read(codec, LLOPM_CTRL); - snd_soc_write(codec, LLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, RLOPM_CTRL); - snd_soc_write(codec, RLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, MONOLOPM_CTRL); - snd_soc_write(codec, MONOLOPM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPLOUT_CTRL); - snd_soc_write(codec, HPLOUT_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPROUT_CTRL); - snd_soc_write(codec, HPROUT_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPLCOM_CTRL); - snd_soc_write(codec, HPLCOM_CTRL, reg | UNMUTE); - reg = snd_soc_read(codec, HPRCOM_CTRL); - snd_soc_write(codec, HPRCOM_CTRL, reg | UNMUTE); + snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE); + snd_soc_update_bits(codec, HPRCOM_CTRL, UNMUTE, UNMUTE); /* ADC default volume and unmute */ snd_soc_write(codec, LADC_VOL, DEFAULT_GAIN); -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC:core:add flag ignore_pmdown_time in snd_soc_dai_link
by ramesh.babu＠linux.intel.com 27 Oct '11

27 Oct '11

With this flag, each dai_link in machine driver can choose to ignore pmdown_time during DAPM shut down sequence. If the ignore_pmdown_time is set, the DAPM for corresponding DAI will be executed immediately. Signed-off-by: Ramesh Babu K V <ramesh.babu(a)linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 3 ++- 2 files changed, 5 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 11cfb59..877fcc1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -718,6 +718,9 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; + /* pmdown_time is ignored at stop */ + unsigned int ignore_pmdown_time:1; + /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ee15337..52a7259 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -319,7 +319,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (unlikely(codec->ignore_pmdown_time)) { + if (unlikely(codec->ignore_pmdown_time || + rtd->dai_link->ignore_pmdown_time)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, codec_dai->driver->playback.stream_name, -- 1.7.4.1

3 2

[alsa-devel] ASoC updates for 3.2
by Mark Brown 27 Oct '11

27 Oct '11

The following changes since commit 226d0f22d044f0151287bb7cf334b85182248f0e: ASoC: keep pointer to resource so it can be freed (2011-10-22 10:46:35 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (2): ASoC: max98095: Convert codec->hw_write to snd_soc_write ASoC: wm8940: Properly set codec->dapm.bias_level sound/soc/codecs/max98095.c | 13 ++++++------- sound/soc/codecs/wm8940.c | 2 ++ 2 files changed, 8 insertions(+), 7 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8940: Properly set codec->dapm.bias_level
by Axel Lin 27 Oct '11

27 Oct '11

Reported-by: Chris Paulson-Ellis <chris(a)edesix.com> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8940.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index fec3892..e8a4ca3 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -488,6 +488,8 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, break; } + codec->dapm.bias_level = level; + return ret; } -- 1.7.5.4

2 4

[alsa-devel] [PATCH 0/13] Convert some MICBIAS widgets to supplies
by Mark Brown 27 Oct '11

27 Oct '11

Since supply widgets are much easier to use than MICBIAS widgets we should really convert existing drivers with MICBIAS widgets to use supply widgets instead. Start this with some easier drivers that don't have any machine drivers. Mark Brown (13): ASoC: Convert wm8400 MICBIAS to a supply widget ASoC: Convert wm8900 MICBIAS to a supply widget ASoC: Convert wm8904 MICBIAS to a supply widget ASoC: Convert wm8960 MICBIAS to a supply widget ASoC: Convert wm8961 MICBIAS to a supply widget ASoC: Convert wm8971 MICBIAS to a supply widget ASoC: Convert wm8974 MICBIAS to a supply widget ASoC: Convert wm8983 MICBIAS to a supply widget ASoC: Convert wm8985 MICBIAS to a supply widget ASoC: Convert wm8988 MICBIAS to a supply widget ASoC: Convert wm8990 MICBIAS to a supply widget ASoC: Convert wm8991 MICBIAS to a supply widget ASoC: Convert wm8995 MICBIASes to supply widgets sound/soc/codecs/wm8400.c | 4 ++-- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8983.c | 3 ++- sound/soc/codecs/wm8985.c | 3 ++- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8990.c | 4 ++-- sound/soc/codecs/wm8991.c | 4 ++-- sound/soc/codecs/wm8995.c | 6 ++++-- 13 files changed, 21 insertions(+), 17 deletions(-)

1 13

[alsa-devel] [PATCH] v2: ASoC: Add new Realtek ALC5632 CODEC driver.
by Leon Romanovsky 26 Oct '11

26 Oct '11

This driver implements basic functionality, using I²C for the control channel. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- v2: Free from checkpatch warnings v1: Initial code drop --- --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/alc5632.c | 1190 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/alc5632.h | 243 +++++++++ 4 files changed, 1439 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/alc5632.c create mode 100644 sound/soc/codecs/alc5632.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..e78ece8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C + select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -169,6 +170,9 @@ config SND_SOC_AK4671 config SND_SOC_ALC5623 tristate +config SND_SOC_ALC5632 + tristate + config SND_SOC_CQ0093VC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..98bf1b6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o +snd-soc-alc5632-objs := alc5632.0 snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -113,6 +114,7 @@ obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c new file mode 100644 index 0000000..96d15b0 --- /dev/null +++ b/sound/soc/codecs/alc5632.c @@ -0,0 +1,1190 @@ +/* +* alc5632.c -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.c by Arnaud Patard +* +* 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. +*/ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/alc5632.h> + +#include "alc5632.h" + +#define ALC5632_VERSION "0.1" + +/* + * ALC5632 register cache + */ +static const u16 alc5632_reg_defaults[] = { + 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ + 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ + 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ + 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ + 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ + 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ + 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ + 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ + 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ + 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ + 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ + 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ + 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ + 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ + 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ + 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ + 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +}; + +static int caps_charge = 2000; +module_param(caps_charge, int, 0); +MODULE_PARM_DESC(caps_charge, "ALC5632 cap charge time (msecs)"); + +/* codec private data */ +struct alc5632_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; + unsigned int add_ctrl; + unsigned int jack_det_ctrl; +}; + +static int alc5632_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case ALC5632_RESET: + case ALC5632_PWR_DOWN_CTRL_STATUS: + case ALC5632_GPIO_PIN_STATUS: + case ALC5632_OVER_CURR_STATUS: + case ALC5632_HID_CTRL_DATA: + case ALC5632_EQ_CTRL: + return 1; + + default: + break; + } + + return 0; +} + +static inline int alc5632_reset(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ALC5632_RESET, 0); + return snd_soc_read(codec, ALC5632_RESET); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5632 Controls + */ + +/* -34.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +/* -46.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +/* -16.5db min scale, 1.5db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +/* 0db min scale, 6 db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); +/* 0db min scalem 0.75db steps, no mute */ +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); + +static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { + /* left starts at bit 8, right at bit 0 */ + /* 31 steps (5 bit), -46.5db scale */ + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + /* bit 15 mutes left, bit 7 right */ + SOC_DOUBLE("Line Playback Switch", + ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5632_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Voice DAC Playback Volume", + ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE_TLV("Phone Capture Volume", + ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Stereo DAC Capture Volume", + ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), + SOC_DOUBLE("Stereo DAC Capture Switch", + ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1), +SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1), +SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1), +SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 9, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1), +SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", + ALC5632_MIC_ROUTING_CTRL, 10, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1), +SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5632_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Mute"}; +static const char *alc5632_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5632_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5632_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5632_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5632_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static const struct snd_kcontrol_new alc5632_auxout_mux_controls = +SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5632_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static const struct snd_kcontrol_new alc5632_spkout_mux_controls = +SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5632_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = +SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5632_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = +SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5632_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = +SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); + +static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5632_hp_mixer_controls[0], + ARRAY_SIZE(alc5632_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0, + &alc5632_hpr_mixer_controls[0], + ARRAY_SIZE(alc5632_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0, + &alc5632_hpl_mixer_controls[0], + ARRAY_SIZE(alc5632_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, + &alc5632_mono_mixer_controls[0], + ARRAY_SIZE(alc5632_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, + &alc5632_speaker_mixer_controls[0], + ARRAY_SIZE(alc5632_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, + &alc5632_captureL_mixer_controls[0], + ARRAY_SIZE(alc5632_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, + &alc5632_captureR_mixer_controls[0], + ARRAY_SIZE(alc5632_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0), + +SND_SOC_DAPM_OUTPUT("AUXOUT"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("PHONEP"), +SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + +static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5632_amp_enum = + SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static const struct snd_kcontrol_new alc5632_amp_mux_controls = + SOC_DAPM_ENUM("Route", alc5632_amp_enum); + +static const struct snd_soc_dapm_widget alc5632_dapm_amp_widgets[] = { +SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, + &alc5632_amp_mux_controls), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"Phone Mix", NULL, "Phone"}, + {"Phone Mix", NULL, "Phone ADMix"}, + {"AUXOUT", NULL, "Aux Out"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + + {"HP Mix", "DACR2HP Playback Switch", "I2S Mix"}, + {"HP Mix", "DACL2HP Playback Switch", "I2S Mix"}, + + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Aux Out", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Phone", NULL, "PHONEP"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"SpeakerOut N Mux", "RN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Right Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + + {"SPKOUT", NULL, "Left Speaker"}, + {"SPKOUT", NULL, "Right Speaker"}, + + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_spk[] = { + {"Right Speaker", NULL, "SpeakerOut Mux"}, + {"Left Speaker", NULL, "SpeakerOut Mux"}, + +}; + +static const struct snd_soc_dapm_route intercon_amp_spk[] = { + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"SpeakerOut", NULL, "AB-D Amp Mux"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5632 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +/* FOUT = MCLK*(N+2)/((M+2)*(K+2)) + N: bit 15:8 (div 2 .. div 257) + K: bit 6:4 typical 2 + M: bit 3:0 (div 2 .. div 17) + + same as for 5623 - thanks! +*/ + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5632_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK) + return -ENODEV; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5632_DAI_CONTROL); + if (reg & ALC5632_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5632_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_BCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5632_PLL_FR_BCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_VBCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from voice clock */ + gbl_clk = ALC5632_PLL_FR_VBCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + +/* choose MCLK/BCLK/VBCLK */ + snd_soc_write(codec, ALC5632_GPCR2, gbl_clk); +/* choose PLL1 clock rate */ + snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div); +/* enable PLL1 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + ALC5632_PWR_ADD2_PLL1); +/* enable PLL2 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + ALC5632_PWR_ADD2_PLL2); +/* use PLL1 as main SYSCLK */ + snd_soc_update_bits(codec, ALC5632_GPCR1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {512*1, 0x3075}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5632->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5632->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5632_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5632_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5632_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5632_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5632_DAI_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5632_DAI_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); +} + +static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5632_DAI_CONTROL); + iface &= ~ALC5632_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5632_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5632_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5632_DAI_I2S_DL_24; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", + __func__, alc5632->sysclk, rate, coeff); + snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff); + + return 0; +} + +static int alc5632_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L \ + |ALC5632_MISC_HP_DEPOP_MUTE_R; + u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg); +} + +#define ALC5632_ADD2_POWER_EN (ALC5632_PWR_ADD2_VREF) + +#define ALC5632_ADD3_POWER_EN (ALC5632_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5632_ADD1_POWER_EN \ + (ALC5632_PWR_ADD1_SPK_AMP_EN \ + | ALC5632_PWR_ADD1_DAC_REF \ + | ALC5632_PWR_ADD1_DAC_L_EN \ + | ALC5632_PWR_ADD1_DAC_R_EN \ + | ALC5632_PWR_ADD1_SOFTGEN_EN \ + | ALC5632_PWR_ADD1_MAIN_I2S_EN \ + | ALC5632_PWR_ADD1_HP_OUT_AMP \ + | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \ + | ALC5632_PWR_ADD1_MAIN_BIAS) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_SOFTGEN_EN, + ALC5632_PWR_ADD1_SOFTGEN_EN); + + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD3, ALC5632_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + ALC5632_MISC_HP_DEPOP_MODE2_EN); + + msleep(500); + + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_ADD2_POWER_EN); + + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, ALC5632_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5632_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_MAIN_BIAS); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_VREF); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, 0); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD3, 0); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5632_dai_ops = { + .hw_params = alc5632_pcm_hw_params, + .digital_mute = alc5632_mute, + .set_fmt = alc5632_set_dai_fmt, + .set_sysclk = alc5632_set_dai_sysclk, + .set_pll = alc5632_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5632_dai = { + .name = "alc5632-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + + .ops = &alc5632_dai_ops, +}; + +static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5632_resume(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) + snd_soc_write(codec, i, cache[i]); + + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* charge alc5632 caps */ + if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->dapm.bias_level = SND_SOC_BIAS_ON; + alc5632_set_bias_level(codec, codec->dapm.bias_level); + } + + return 0; +} + +#define ALC5632_REC_UNMUTE (ALC5632_ADC_REC_MIC2 \ + | ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX \ + | ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK \ + | ALC5632_ADC_REC_MONOMIX) + +#define ALC5632_MIC_ROUTE (ALC5632_MIC_ROUTE_HP \ + | ALC5632_MIC_ROUTE_SPK \ + | ALC5632_MIC_ROUTE_MONOMIX) + +#define ALC5632_PWR_DEFAULT (ALC5632_PWR_ADC_STATUS \ + | ALC5632_PWR_DAC_STATUS \ + | ALC5632_PWR_AMIX_STATUS \ + | ALC5632_PWR_VREF_STATUS) + +#define ALC5632_ADC_REC_GAIN_COMP(x) (int)((x - ALC5632_ADC_REC_GAIN_BASE) \ + / ALC5632_ADC_REC_GAIN_STEP) + +#define ALC5632_MIC_BOOST_COMP(x) (int)(x / ALC5632_MIC_BOOST_STEP) + +#define ALC5632_SPK_OUT_VOL_COMP(x) (int)(x / ALC5632_SPK_OUT_VOL_STEP) + +static int alc5632_probe(struct snd_soc_codec *codec) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + dev_dbg(codec->dev, "ALC5632 Audio Codec Ver. %s\n", ALC5632_VERSION); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5632_reset(codec); + + /* power on device */ + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (alc5632->add_ctrl) { + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, + alc5632->add_ctrl); + } + + /* spk amp pwr enable 3A | 0x0400 @ 3A */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + 0, ALC5632_PWR_ADD1_SPK_AMP_EN); + + /* "normal" mode: 0 @ 26 */ + snd_soc_write(codec, ALC5632_PWR_DOWN_CTRL_STATUS, 0); + + /* power on VREF on all analog circuits 0x2000 @ 3C */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + 0, ALC5632_PWR_ADD2_VREF); + + /* enable slave mode 0x8000 @ 34 */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, ALC5632_DAI_SDP_SLAVE_MODE); + + switch (alc5632->id) { + case 0x5c: + snd_soc_add_controls(codec, alc5632_vol_snd_controls, + ARRAY_SIZE(alc5632_vol_snd_controls)); + break; + default: + return -EINVAL; + } + + snd_soc_add_controls(codec, alc5632_snd_controls, + ARRAY_SIZE(alc5632_snd_controls)); + + snd_soc_dapm_new_controls(dapm, alc5632_dapm_widgets, + ARRAY_SIZE(alc5632_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + return ret; +} + +/* power down chip */ +static int alc5632_remove(struct snd_soc_codec *codec) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5632 = { + .probe = alc5632_probe, + .remove = alc5632_remove, + .suspend = alc5632_suspend, + .resume = alc5632_resume, + .set_bias_level = alc5632_set_bias_level, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = alc5632_reg_defaults, + .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_register = alc5632_volatile_register, +}; + +/* + * alc5632 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5632_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5632_platform_data *pdata; + struct alc5632_priv *alc5632; + int ret, vid1, vid2; + + dev_dbg(&client->dev, "in i2c probe...\n"); + + vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_err(&client->dev, "got vid1: %x\n", vid1); + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_err(&client->dev, "got vid2: %x\n", vid2); + } + vid2 = (vid2 & 0xff); + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + dev_dbg(&client->dev, "Found codec: ALC5632 (ID: %x)\n", vid2); + + alc5632 = kzalloc(sizeof(struct alc5632_priv), GFP_KERNEL); + if (alc5632 == NULL) + return -ENOMEM; + + pdata = client->dev.platform_data; + if (pdata) { + alc5632->add_ctrl = pdata->add_ctrl; + alc5632->jack_det_ctrl = pdata->jack_det_ctrl; + } + + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + kfree(alc5632); + return -EINVAL; + } + + i2c_set_clientdata(client, alc5632); + alc5632->control_data = client; + alc5632->control_type = SND_SOC_I2C; + mutex_init(&alc5632->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5632, &alc5632_dai, 1); + if (ret != 0) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(alc5632); + } + + return ret; +} + +static int alc5632_i2c_remove(struct i2c_client *client) +{ + struct alc5632_priv *alc5632 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(alc5632); + return 0; +} + +static const struct i2c_device_id alc5632_i2c_table[] = { + {"alc5632", 0x5c}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5632_i2c_driver = { + .driver = { + .name = "alc5632", + .owner = THIS_MODULE, + }, + .probe = alc5632_i2c_probe, + .remove = __devexit_p(alc5632_i2c_remove), + .id_table = alc5632_i2c_table, +}; + +static int __init alc5632_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5632_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5632_modinit); + +static void __exit alc5632_modexit(void) +{ + i2c_del_driver(&alc5632_i2c_driver); +} +module_exit(alc5632_modexit); + +MODULE_DESCRIPTION("ASoC ALC5632 driver"); +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h new file mode 100644 index 0000000..9747016 --- /dev/null +++ b/sound/soc/codecs/alc5632.h @@ -0,0 +1,243 @@ +/* +* alc5632.h -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.h by Arnaud Patard +* +* 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 _ALC5632_H +#define _ALC5632_H + +#define ALC5632_RESET 0x00 +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5632_SPK_OUT_VOL 0x02 /* spe out vol */ +#define ALC5632_SPK_OUT_VOL_STEP 1.5 +#define ALC5632_HP_OUT_VOL 0x04 /* hp out vol */ +#define ALC5632_AUX_OUT_VOL 0x06 /* aux out vol */ +#define ALC5632_PHONE_IN_VOL 0x08 /* phone in vol */ +#define ALC5632_LINE_IN_VOL 0x0A /* line in vol */ +#define ALC5632_STEREO_DAC_IN_VOL 0x0C /* stereo dac in vol */ +#define ALC5632_MIC_VOL 0x0E /* mic in vol */ +/* stero dac/mic routing */ +#define ALC5632_MIC_ROUTING_CTRL 0x10 +#define ALC5632_MIC_ROUTE_MONOMIX (1 << 0) +#define ALC5632_MIC_ROUTE_SPK (1 << 1) +#define ALC5632_MIC_ROUTE_HP (1 << 2) + +#define ALC5632_ADC_REC_GAIN 0x12 /* rec gain */ +#define ALC5632_ADC_REC_GAIN_RANGE 0x1F1F +#define ALC5632_ADC_REC_GAIN_BASE (-16.5) +#define ALC5632_ADC_REC_GAIN_STEP 1.5 + +#define ALC5632_ADC_REC_MIXER 0x14 /* mixer control */ +#define ALC5632_ADC_REC_MIC1 (1 << 6) +#define ALC5632_ADC_REC_MIC2 (1 << 5) +#define ALC5632_ADC_REC_LINE_IN (1 << 4) +#define ALC5632_ADC_REC_AUX (1 << 3) +#define ALC5632_ADC_REC_HP (1 << 2) +#define ALC5632_ADC_REC_SPK (1 << 1) +#define ALC5632_ADC_REC_MONOMIX (1 << 0) + +#define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */ +/* ALC5632_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */ +#define ALC5632_OUTPUT_MIXER_RP (1 << 14) +#define ALC5632_OUTPUT_MIXER_WEEK (1 << 12) +#define ALC5632_OUTPUT_MIXER_HP (1 << 10) +#define ALC5632_OUTPUT_MIXER_AUX_SPK (2 << 6) +#define ALC5632_OUTPUT_MIXER_AUX_HP_LR (1 << 6) +#define ALC5632_OUTPUT_MIXER_HP_R (1 << 8) +#define ALC5632_OUTPUT_MIXER_HP_L (1 << 9) + +#define ALC5632_MIC_CTRL 0x22 /* mic phone ctrl */ +#define ALC5632_MIC_BOOST_BYPASS 0 +#define ALC5632_MIC_BOOST_20DB 1 +#define ALC5632_MIC_BOOST_30DB 2 +#define ALC5632_MIC_BOOST_40DB 3 + +#define ALC5632_DIGI_BOOST_CTRL 0x24 /* digi mic / bost ctl */ +#define ALC5632_MIC_BOOST_RANGE 7 +#define ALC5632_MIC_BOOST_STEP 6 +#define ALC5632_PWR_DOWN_CTRL_STATUS 0x26 +#define ALC5632_PWR_VREF_STATUS (1 << 3) +#define ALC5632_PWR_AMIX_STATUS (1 << 2) +#define ALC5632_PWR_DAC_STATUS (1 << 1) +#define ALC5632_PWR_ADC_STATUS (1 << 0) +/* stereo/voice DAC / stereo adc func ctrl */ +#define ALC5632_DAC_FUNC_SELECT 0x2E + +/* Main serial data port ctrl (i2s) */ +#define ALC5632_DAI_CONTROL 0x34 + +#define ALC5632_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5632_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5632_DAI_SADLRCK_MODE (1 << 14) +/* 0:voice, 1:main */ +#define ALC5632_DAI_MAIN_I2S_SYSCLK_SEL (1 << 8) +#define ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_MAIN_I2S_LRCK_INV (1 << 6) +#define ALC5632_DAI_I2S_DL_MASK (3 << 2) +#define ALC5632_DAI_I2S_DL_8 (3 << 2) +#define ALC5632_DAI_I2S_DL_24 (2 << 2) +#define ALC5632_DAI_I2S_DL_20 (1 << 2) +#define ALC5632_DAI_I2S_DL_16 (0 << 2) +#define ALC5632_DAI_I2S_DF_MASK (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_B (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_A (2 << 0) +#define ALC5632_DAI_I2S_DF_LEFT (1 << 0) +#define ALC5632_DAI_I2S_DF_I2S (0 << 0) +/* extend serial data port control (VoDAC_i2c/pcm) */ +#define ALC5632_DAI_CONTROL2 0x36 +/* 0:gpio func, 1:voice pcm */ +#define ALC5632_DAI_VOICE_PCM_ENABLE (1 << 15) +/* 0:master, 1:slave */ +#define ALC5632_DAI_VOICE_MODE_SEL (1 << 14) +/* 0:disable, 1:enable */ +#define ALC5632_DAI_HPF_CLK_CTRL (1 << 13) +/* 0:main, 1:voice */ +#define ALC5632_DAI_VOICE_I2S_SYSCLK_SEL (1 << 8) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_VBCLK_SYSCLK_SEL (1 << 7) +/* 0:normal, 1:invert */ +#define ALC5632_DAI_VOICE_I2S_LR_INV (1 << 6) +#define ALC5632_DAI_VOICE_DL_MASK (3 << 2) +#define ALC5632_DAI_VOICE_DL_16 (0 << 2) +#define ALC5632_DAI_VOICE_DL_20 (1 << 2) +#define ALC5632_DAI_VOICE_DL_24 (2 << 2) +#define ALC5632_DAI_VOICE_DL_8 (3 << 2) +#define ALC5632_DAI_VOICE_DF_MASK (3 << 0) +#define ALC5632_DAI_VOICE_DF_I2S (0 << 0) +#define ALC5632_DAI_VOICE_DF_LEFT (1 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_A (2 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_B (3 << 0) + +#define ALC5632_PWR_MANAG_ADD1 0x3A +#define ALC5632_PWR_ADD1_DAC_L_EN (1 << 15) +#define ALC5632_PWR_ADD1_DAC_R_EN (1 << 14) +#define ALC5632_PWR_ADD1_ZERO_CROSS (1 << 13) +#define ALC5632_PWR_ADD1_MAIN_I2S_EN (1 << 11) +#define ALC5632_PWR_ADD1_SPK_AMP_EN (1 << 10) +#define ALC5632_PWR_ADD1_HP_OUT_AMP (1 << 9) +#define ALC5632_PWR_ADD1_HP_OUT_ENH_AMP (1 << 8) +#define ALC5632_PWR_ADD1_VOICE_DAC_MIX (1 << 7) +#define ALC5632_PWR_ADD1_SOFTGEN_EN (1 << 6) +#define ALC5632_PWR_ADD1_MIC1_SHORT_CURR (1 << 5) +#define ALC5632_PWR_ADD1_MIC2_SHORT_CURR (1 << 4) +#define ALC5632_PWR_ADD1_MIC1_EN (1 << 3) +#define ALC5632_PWR_ADD1_MIC2_EN (1 << 2) +#define ALC5632_PWR_ADD1_MAIN_BIAS (1 << 1) +#define ALC5632_PWR_ADD1_DAC_REF (1 << 0) + +#define ALC5632_PWR_MANAG_ADD2 0x3C +#define ALC5632_PWR_ADD2_PLL1 (1 << 15) +#define ALC5632_PWR_ADD2_PLL2 (1 << 14) +#define ALC5632_PWR_ADD2_VREF (1 << 13) +#define ALC5632_PWR_ADD2_OVT_DET (1 << 12) +#define ALC5632_PWR_ADD2_VOICE_DAC (1 << 10) +#define ALC5632_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5632_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5632_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5632_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5632_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5632_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5632_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5632_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5632_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5632_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5632_PWR_MANAG_ADD3 0x3E +#define ALC5632_PWR_ADD3_AUXOUT_VOL (1 << 14) +#define ALC5632_PWR_ADD3_SPK_L_OUT (1 << 13) +#define ALC5632_PWR_ADD3_SPK_R_OUT (1 << 12) +#define ALC5632_PWR_ADD3_HP_L_OUT_VOL (1 << 11) +#define ALC5632_PWR_ADD3_HP_R_OUT_VOL (1 << 10) +#define ALC5632_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5632_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5632_PWR_ADD3_AUXIN_VOL (1 << 5) +#define ALC5632_PWR_ADD3_AUXIN_MIX (1 << 4) +#define ALC5632_PWR_ADD3_MIC1_VOL (1 << 3) +#define ALC5632_PWR_ADD3_MIC2_VOL (1 << 2) +#define ALC5632_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5632_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5632_GPCR1 0x40 +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1 (1 << 15) +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_MCLK (0 << 15) +#define ALC5632_GPCR1_DAC_HI_FLT_EN (1 << 10) +#define ALC5632_GPCR1_SPK_AMP_CTRL (7 << 1) +#define ALC5632_GPCR1_VDD_100 (5 << 1) +#define ALC5632_GPCR1_VDD_125 (4 << 1) +#define ALC5632_GPCR1_VDD_150 (3 << 1) +#define ALC5632_GPCR1_VDD_175 (2 << 1) +#define ALC5632_GPCR1_VDD_200 (1 << 1) +#define ALC5632_GPCR1_VDD_225 (0 << 1) + +#define ALC5632_GPCR2 0x42 +#define ALC5632_GPCR2_PLL1_SOUR_SEL (3 << 12) +#define ALC5632_PLL_FR_MCLK (0 << 12) +#define ALC5632_PLL_FR_BCLK (2 << 12) +#define ALC5632_PLL_FR_VBCLK (3 << 12) +#define ALC5632_GPCR2_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5632_PLL1_CTRL 0x44 +#define ALC5632_PLL1_CTRL_N_VAL(n) (((n) & 0x0f) << 8) +#define ALC5632_PLL1_M_BYPASS (1 << 7) +#define ALC5632_PLL1_CTRL_K_VAL(k) (((k) & 0x07) << 4) +#define ALC5632_PLL1_CTRL_M_VAL(m) (((m) & 0x0f) << 0) + +#define ALC5632_PLL2_CTRL 0x46 +#define ALC5632_PLL2_EN (1 << 15) +#define ALC5632_PLL2_RATIO (0 << 15) + +#define ALC5632_GPIO_PIN_CONFIG 0x4C +#define ALC5632_GPIO_PIN_POLARITY 0x4E +#define ALC5632_GPIO_PIN_STICKY 0x50 +#define ALC5632_GPIO_PIN_WAKEUP 0x52 +#define ALC5632_GPIO_PIN_STATUS 0x54 +#define ALC5632_GPIO_PIN_SHARING 0x56 +#define ALC5632_OVER_CURR_STATUS 0x58 +#define ALC5632_SOFTVOL_CTRL 0x5A +#define ALC5632_GPIO_OUPUT_PIN_CTRL 0x5C + +#define ALC5632_MISC_CTRL 0x5E +#define ALC5632_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5632_MISC_AVC_TRGT_SEL (3 << 12) +#define ALC5632_MISC_AVC_TRGT_RIGHT (1 << 12) +#define ALC5632_MISC_AVC_TRGT_LEFT (2 << 12) +#define ALC5632_MISC_AVC_TRGT_BOTH (3 << 12) +#define ALC5632_MISC_HP_DEPOP_MODE1_EN (1 << 9) +#define ALC5632_MISC_HP_DEPOP_MODE2_EN (1 << 8) +#define ALC5632_MISC_HP_DEPOP_MUTE_L (1 << 7) +#define ALC5632_MISC_HP_DEPOP_MUTE_R (1 << 6) +#define ALC5632_MISC_HP_DEPOP_MUTE (1 << 5) +#define ALC5632_MISC_GPIO_WAKEUP_CTRL (1 << 1) +#define ALC5632_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5632_DAC_CLK_CTRL1 0x60 +#define ALC5632_DAC_CLK_CTRL2 0x62 +#define ALC5632_DAC_CLK_CTRL2_DIV1_2 (1 << 0) +#define ALC5632_VOICE_DAC_PCM_CLK_CTRL1 0x64 +#define ALC5632_PSEUDO_SPATIAL_CTRL 0x68 +#define ALC5632_HID_CTRL_INDEX 0x6A +#define ALC5632_HID_CTRL_DATA 0x6C +#define ALC5632_EQ_CTRL 0x6E + +/* undocumented */ +#define ALC5632_VENDOR_ID1 0x7C +#define ALC5632_VENDOR_ID2 0x7E + +#endif -- 1.7.3.4

2 1

Re: [alsa-devel] [PATCH 1/1] intel8x0: [v2] Improve performance in virtual environment
by Takashi Iwai 26 Oct '11

26 Oct '11

At Wed, 26 Oct 2011 17:29:37 +0400, Konstantin Ozerkov wrote: > > On 26.10.11 17:03, Takashi Iwai wrote: > > Are the changes specific to x86, right? > > boot_cpu_has() seems x86 only, and kvm_para_*() isn't provided for all > > architectures. > > > > The best would be to provide a standard API for such detection. But, > > as a workaround, we may need a simple ifdef. > Is this really needed ? This is driver for controllers which is used only (AFIK) with x86 and x86_64 CPUs. I remember vaguely the implementation on PPC and others. Who knows. (And you can run a kernel in VM, right? :) Takashi

1 0

[alsa-devel] ASoC PCM drivers without progress counters breaks snd_pcm_status
by Magnus Olsson 26 Oct '11

26 Oct '11

Hi, To determine the timestamp of captured audio, we do something along the lines: ts = snd_pcm_status_get_htstamp() - time(snd_pcm_status_get_delay()) snd_pcm_readi(buf) process(buf, ts) However for PCM drivers that only update its hw_pointer in steps of periods (in irq-handler), the attributes in snd_pcm_status() struct will not be in sync even though ALSA thinks so. For example, the avail property will correspond to the time of the last IRQ, not at the specified timestamp returned by snd_pcm_status_get_htstamp(). This will introduce timestamp jitter which depends on system IRQ latency and also when the userspace application issues it syscall to read status. For an embedded system under heavy load, the A/V sync is unacceptable when using ALSA timestamps. Several mainline PCM drivers do this kind of hw-pointer caching in their irq handlers, perhaps because the DMA HW does not support progress counting. Possible solutions: 1) If DMA HW supports progress counter, implement it. 2) If DMA HW is not supported, implement it in software (estimate position using elapsed time since last irq). Only requires driver changes. 3) If DMA HW is not supported, return use last irq timestamp as the timestamp returned by snd_pcm_status_get_htstamp() . Probably requires ALSA Core changes. I've tested 1) and results in much improved A/V sync on our systems. If would be kind of convenient if ALSA SoC Core could provide a generic pointercallback for solution 2), if the PCM driver specifies that it does not support HW progress counter. Otherwise we will see a lot of duplicate code for estimating position. AFAIK, ALSA already maintains a timestamp for last irq. Magnus Olsson Axis Communications / Product Platforms

2 1

Re: [alsa-devel] [PATCH 1/1] intel8x0: [v2] Improve performance in virtual environment
by Takashi Iwai 26 Oct '11

26 Oct '11

At Wed, 26 Oct 2011 16:00:15 +0400, Konstantin Ozerkov wrote: > > v2: add detection for virtual environments (KVM and Parallels) > > This patch is intended to improve performance in virtualized environments > like Parallels Desktop or KVM/VirtualBox/QEMU (virtual ICH/AC97 audio). > > I/O access is very time-expensive operation in virtual world: VCPU > can be rescheduled and in the worst case we get more than 10ms delay on > each I/O access. > > In the virtual environment loop exit rule > (old_civ == current_civ && old_picb == current_picb) is never satisfied, > because old_picb is never the same as current_picb due to delay inspired > by reading current_civ. As a result loop ended by timeout and we get 10x > more I/O operations. > > Experimental data from Prallels Desktop 7, RHEL6 guest (I/O ops per > second): > > Original code: > In Port Counter Callback > f014 41550 fffff00000179d00 ac97_bm_read_civ+0x000 > f018 41387 fffff0000017a580 ac97_bm_read_picb+0x000 > > With patch: > In Port Counter Callback > f014 4090 fffff00000179d00 ac97_bm_read_civ+0x000 > f018 1964 fffff0000017a580 ac97_bm_read_picb+0x000 > > Signed-off-by: Konstantin Ozerkov <kozerkov(a)parallels.com> > Signed-off-by: Denis V. Lunev <den(a)openvz.org> Are the changes specific to x86, right? boot_cpu_has() seems x86 only, and kvm_para_*() isn't provided for all architectures. The best would be to provide a standard API for such detection. But, as a workaround, we may need a simple ifdef. thanks, Takashi > diff --git a/pci/intel8x0.c b/pci/intel8x0.c > index 6a5b387..0974ae5 100644 > --- a/pci/intel8x0.c > +++ b/pci/intel8x0.c > @@ -27,6 +27,7 @@ > */ > > #include <asm/io.h> > +#include <asm/kvm_para.h> > #include <linux/delay.h> > #include <linux/interrupt.h> > #include <linux/init.h> > @@ -77,6 +78,7 @@ static int buggy_semaphore; > static int buggy_irq = -1; /* auto-check */ > static int xbox; > static int spdif_aclink = -1; > +static int inside_vm = -1; > > module_param(index, int, 0444); > MODULE_PARM_DESC(index, "Index value for Intel i8x0 soundcard."); > @@ -94,6 +96,8 @@ module_param(xbox, bool, 0444); > MODULE_PARM_DESC(xbox, "Set to 1 for Xbox, if you have problems with the AC'97 codec detection."); > module_param(spdif_aclink, int, 0444); > MODULE_PARM_DESC(spdif_aclink, "S/PDIF over AC-link."); > +module_param(inside_vm, bool, 0444); > +MODULE_PARM_DESC(inside_vm, "KVM/Parallels optimization."); > > /* just for backward compatibility */ > static int enable; > @@ -400,6 +404,7 @@ struct intel8x0 { > unsigned buggy_irq: 1; /* workaround for buggy mobos */ > unsigned xbox: 1; /* workaround for Xbox AC'97 detection */ > unsigned buggy_semaphore: 1; /* workaround for buggy codec semaphore */ > + unsigned inside_vm: 1; /* enable VM optimization */ > > int spdif_idx; /* SPDIF BAR index; *_SPBAR or -1 if use PCMOUT */ > unsigned int sdm_saved; /* SDM reg value */ > @@ -1065,8 +1070,11 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs > udelay(10); > continue; > } > - if (civ == igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV) && > - ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb)) > + if (civ != igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV)) > + continue; > + if (chip->inside_vm) > + break; > + if (ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb)) > break; > } while (timeout--); > ptr = ichdev->last_pos; > @@ -2984,6 +2992,10 @@ static int __devinit snd_intel8x0_create(struct snd_card *card, > if (xbox) > chip->xbox = 1; > > + chip->inside_vm = inside_vm; > + if (inside_vm) > + printk(KERN_INFO "intel8x0: enable KVM optimization\n"); > + > if (pci->vendor == PCI_VENDOR_ID_INTEL && > pci->device == PCI_DEVICE_ID_INTEL_440MX) > chip->fix_nocache = 1; /* enable workaround */ > @@ -3226,6 +3238,11 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci, > buggy_irq = 0; > } > > + if (inside_vm < 0) { > + /* detect KVM and Parallels virtual environments */ > + inside_vm = kvm_para_available() || boot_cpu_has(X86_FEATURE_HYPERVISOR); > + } > + > if ((err = snd_intel8x0_create(card, pci, pci_id->driver_data, > &chip)) < 0) { > snd_card_free(card); > -- > 1.7.7 >

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Re: [alsa-devel] [PATCH] hda_hwdep: Fix possible buffer overflow
by Takashi Iwai 26 Oct '11

26 Oct '11

At Wed, 26 Oct 2011 09:48:12 +0200, Alexander Stein wrote: > > If a line in the firmware file is larger than the given buffer size (and > so the firmware file size), size is set to a value larger than the actual > buffer size. This results in an overflow in the buffer passed. > Fix this by copying only up to 127 chars per line. Actually this check should have been if (size > fw->size) size = fw->size; Otherwise it doesn't make sense. If the change is OK, could you resend the patch with it? thanks, Takashi > Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> > --- > sound/pci/hda/hda_hwdep.c | 6 +++--- > 1 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c > index bf3ced5..61da08d 100644 > --- a/sound/pci/hda/hda_hwdep.c > +++ b/sound/pci/hda/hda_hwdep.c > @@ -745,6 +745,7 @@ static int parse_line_mode(char *buf, struct hda_bus *bus) > * if successfully copied a line > * > * the spaces at the beginning and the end of the line are stripped > + * lines read are clamped to 127 chars > */ > static int get_line_from_fw(char *buf, int size, struct firmware *fw) > { > @@ -756,8 +757,6 @@ static int get_line_from_fw(char *buf, int size, struct firmware *fw) > } > if (!fw->size) > return 0; > - if (size < fw->size) > - size = fw->size; > > for (len = 0; len < fw->size; len++) { > if (!*p) > @@ -768,7 +767,8 @@ static int get_line_from_fw(char *buf, int size, struct firmware *fw) > break; > } > if (len < size) > - *buf++ = *p++; > + *buf++ = *p; > + p++; > } > *buf = 0; > fw->size -= len; > -- > 1.7.3.4 >

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[alsa-devel] alsa lib bug in 1.0.24
by Scott Jiang 26 Oct '11

26 Oct '11

Hi Takashi and Mark, When I use ad1836 and ad1938 with alsa lib 1.0.24, there is an error: root:/> arecord -t wav -c 2 -d 3 -r 48000 -f S32_LE 1.wav Recording WAVE '1.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo arecord: pcm_read:1773: read error: Input/output error If I use S16_LE and S24_LE there is no error. The drivers for ad1836 and ad1938 only support S32_LE. My card config file lists below: bfin-ad1836.pcm.default { @args [ CARD ] @args.CARD { type string } type plug slave.pcm { type mmap_emul slave.pcm { type hw card $CARD format S32_LE rate 48000 } } } If I used alsa lib 1.0.18, S16_LE, S24_LE and S32_LE are all ok. Is my config file wrong in new version lib? It seems if the format is the same as driver supports it will get an error. Regards, Scott

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Re: [alsa-devel] dmix - unable to open slave
by Scott Jiang 26 Oct '11

26 Oct '11

2011/10/25 Nenad Sljivic <nenad.sljivic(a)rt-rk.com>: > Hello Scott, > > Thanks a lot for your reply. My project is totally blocked due to this > issue. I would really appreciate your help. > > Are you referring to this entry: > http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/043393.h > tml > ? > > Could you elaborate your solution a little more ? > > What do you mean by: > " I add driver_name in my snd_soc_card to solve this bug. " > Mark has committed a patch to fix this in asoc core. you can cat /proc/asound/cards to find the card driver name, then you must modify your config file to match this. Scott

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[alsa-devel] [PATCH] ASoC: Add new Realtek ALC5632 CODEC driver.
by Leon Romanovsky 25 Oct '11

25 Oct '11

This driver implements basic functionality, using I²C for the control channel. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/alc5632.c | 1175 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/alc5632.h | 233 +++++++++ 4 files changed, 1414 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/alc5632.c create mode 100644 sound/soc/codecs/alc5632.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..e78ece8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C select SND_SOC_ALC5623 if I2C + select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -169,6 +170,9 @@ config SND_SOC_AK4671 config SND_SOC_ALC5623 tristate +config SND_SOC_ALC5632 + tristate + config SND_SOC_CQ0093VC tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..98bf1b6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -29,6 +29,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o +snd-soc-alc5632-objs := alc5632.0 snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o @@ -113,6 +114,7 @@ obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c new file mode 100644 index 0000000..b3670a3 --- /dev/null +++ b/sound/soc/codecs/alc5632.c @@ -0,0 +1,1175 @@ +/* +* alc5632.c -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.c by Arnaud Patard +* +* 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. +*/ + +/* #define DEBUG */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/alc5632.h> + +#include "alc5632.h" + +#define ALC5632_VERSION "0.1" + +/* + * ALC5632 register cache + */ +static const u16 alc5632_reg_defaults[] = { + 0x59B4, 0x0000, 0x8080, 0x0000, /* 0 */ + 0x8080, 0x0000, 0x8080, 0x0000, /* 4 */ + 0xC800, 0x0000, 0xE808, 0x0000, /* 8 */ + 0x1010, 0x0000, 0x0808, 0x0000, /* 12 */ + 0xEE0F, 0x0000, 0xCBCB, 0x0000, /* 16 */ + 0x7F7F, 0x0000, 0x0000, 0x0000, /* 20 */ + 0xE010, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x8008, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 32 */ + 0x00C0, 0x0000, 0xEF00, 0x0000, /* 36 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 44 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 48 */ + 0x8000, 0x0000, 0x0000, 0x0000, /* 52 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */ + 0x0000, 0x0000, 0x8000, 0x0000, /* 60 */ + 0x0C0A, 0x0000, 0x0000, 0x0000, /* 64 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 68 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 72 */ + 0xBE3E, 0x0000, 0xBE3E, 0x0000, /* 76 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 80 */ + 0x803A, 0x0000, 0x0000, 0x0000, /* 84 */ + 0x0000, 0x0000, 0x0009, 0x0000, /* 88 */ + 0x0000, 0x0000, 0x3000, 0x0000, /* 92 */ + 0x3075, 0x0000, 0x1010, 0x0000, /* 96 */ + 0x3110, 0x0000, 0x0000, 0x0000, /* 100 */ + 0x0553, 0x0000, 0x0000, 0x0000, /* 104 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 108 */ +}; + +static int caps_charge = 2000; +module_param(caps_charge, int, 0); +MODULE_PARM_DESC(caps_charge, "ALC5632 cap charge time (msecs)"); + +/* codec private data */ +struct alc5632_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; + unsigned int add_ctrl; + unsigned int jack_det_ctrl; +}; + +static int alc5632_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +{ + switch (reg) { + case ALC5632_RESET: + case ALC5632_PWR_DOWN_CTRL_STATUS: + case ALC5632_GPIO_PIN_STATUS: + case ALC5632_OVER_CURR_STATUS: + case ALC5632_HID_CTRL_DATA: + case ALC5632_EQ_CTRL: + return 1; + + default: + break; + } + + return 0; +} + +static inline int alc5632_reset(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, ALC5632_RESET, 0); + return snd_soc_read(codec, ALC5632_RESET); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5632_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5632_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5632 Controls + */ + +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); // -34.5db min scale, 1.5db steps, no mute +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); // -46.5db min scale, 1.5db steps, no mute +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); // -16.5db min scale, 1.5db steps, no mute +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); // 0db min scale, 6 db steps, no mute +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); // 0db min scalem 0.75db steps, no mute + +static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5632_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), // left starts at bit 8, right at bit 0 + // 31 steps (5 bit), -46.5db scale + SOC_DOUBLE("Line Playback Switch", + ALC5632_SPK_OUT_VOL, 15, 7, 1, 1), // bit 15 mutes left, bit 7 right + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5632_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5632_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5632_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Voice DAC Playback Volume", + ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE_TLV("Phone Capture Volume", + ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Stereo DAC Capture Volume", + ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), + SOC_DOUBLE("Stereo DAC Capture Switch", + ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5632_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5632_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("PHONE2HP Playback Switch", ALC5632_PHONE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 11, 1, 1), +SOC_DAPM_SINGLE("DACL2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 3, 1, 1), +SOC_DAPM_SINGLE("DACR2HP Playback Switch", ALC5632_MIC_ROUTING_CTRL, 2, 1, 1), +SOC_DAPM_SINGLE("VOICE2HP Playback Switch", ALC5632_VOICE_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5632_ADC_REC_GAIN, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5632_ADC_REC_GAIN, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5632_ADC_REC_GAIN, 14, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5632_ADC_REC_GAIN, 6, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5632_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 9, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5632_MIC_ROUTING_CTRL, 0, 1, 1), +SOC_DAPM_SINGLE("VOICE2MONO Playback Switch", ALC5632_VOICE_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5632_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5632_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("PHONE2SPK Playback Switch", ALC5632_PHONE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 10, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5632_MIC_ROUTING_CTRL, 1, 1, 1), +SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5632_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Mute"}; +static const char *alc5632_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5632_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5632_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5632_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5632_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static const struct snd_kcontrol_new alc5632_auxout_mux_controls = +SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5632_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static const struct snd_kcontrol_new alc5632_spkout_mux_controls = +SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5632_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = +SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5632_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = +SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5632_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = +SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); + +static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5632_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5632_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5632_hp_mixer_controls[0], + ARRAY_SIZE(alc5632_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5632_PWR_MANAG_ADD2, 4, 0, + &alc5632_hpr_mixer_controls[0], + ARRAY_SIZE(alc5632_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5632_PWR_MANAG_ADD2, 5, 0, + &alc5632_hpl_mixer_controls[0], + ARRAY_SIZE(alc5632_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, + &alc5632_mono_mixer_controls[0], + ARRAY_SIZE(alc5632_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, + &alc5632_speaker_mixer_controls[0], + ARRAY_SIZE(alc5632_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, + &alc5632_captureL_mixer_controls[0], + ARRAY_SIZE(alc5632_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, + &alc5632_captureR_mixer_controls[0], + ARRAY_SIZE(alc5632_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", ALC5632_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Aux Out", ALC5632_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5632_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5632_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone", ALC5632_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Phone ADMix", ALC5632_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5632_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5632_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5632_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5632_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5632_PWR_MANAG_ADD1, 3, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias2", ALC5632_PWR_MANAG_ADD1, 2, 0), + +SND_SOC_DAPM_OUTPUT("AUXOUT"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("PHONEP"), +SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + +static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5632_amp_enum = + SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static const struct snd_kcontrol_new alc5632_amp_mux_controls = + SOC_DAPM_ENUM("Route", alc5632_amp_enum); + +static const struct snd_soc_dapm_widget alc5632_dapm_amp_widgets[] = { +SND_SOC_DAPM_PGA_E("D Amp", ALC5632_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5632_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, + &alc5632_amp_mux_controls), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"Phone Mix", NULL, "Phone"}, + {"Phone Mix", NULL, "Phone ADMix"}, + {"AUXOUT", NULL, "Aux Out"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + + {"HP Mix", "DACR2HP Playback Switch", "I2S Mix"}, + {"HP Mix", "DACL2HP Playback Switch", "I2S Mix"}, + + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "PHONE2SPK Playback Switch", "Phone Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Aux Out", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Phone", NULL, "PHONEP"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + {"SpeakerOut N Mux", "RN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "RP/+R", "Right Speaker"}, + {"SpeakerOut N Mux", "LN/-R", "Right Speaker"}, + {"SpeakerOut N Mux", "Mute", "Vmid"}, + + + {"SPKOUT", NULL, "Left Speaker"}, + {"SPKOUT", NULL, "Right Speaker"}, + + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_spk[] = { + {"Right Speaker", NULL, "SpeakerOut Mux"}, + {"Left Speaker", NULL, "SpeakerOut Mux"}, + +}; + +static const struct snd_soc_dapm_route intercon_amp_spk[] = { + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"SpeakerOut", NULL, "AB-D Amp Mux"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5632 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +/* FOUT = MCLK*(N+2)/((M+2)*(K+2)) + N: bit 15:8 (div 2 .. div 257) + K: bit 6:4 typical 2 + M: bit 3:0 (div 2 .. div 17) + + same as for 5623 - thanks! +*/ + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5632_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5632_PLL_FR_MCLK || pll_id > ALC5632_PLL_FR_VBCLK) + return -ENODEV; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + 0); + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5632_DAI_CONTROL); + if (reg & ALC5632_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5632_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_BCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5632_PLL_FR_BCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + case ALC5632_PLL_FR_VBCLK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from voice clock */ + gbl_clk = ALC5632_PLL_FR_VBCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + +/* choose MCLK/BCLK/VBCLK */ + snd_soc_write(codec, ALC5632_GPCR2, gbl_clk); +/* choose PLL1 clock rate */ + snd_soc_write(codec, ALC5632_PLL1_CTRL, pll_div); +/* enable PLL1 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL1, + ALC5632_PWR_ADD2_PLL1); +/* enable PLL2 */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_PLL2, + ALC5632_PWR_ADD2_PLL2); +/* use PLL1 as main SYSCLK */ + snd_soc_update_bits(codec, ALC5632_GPCR1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1, + ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {512*1, 0x3075}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5632->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5632->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5632_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5632_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5632_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5632_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5632_DAI_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5632_DAI_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); +} + +static int alc5632_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5632_DAI_CONTROL); + iface &= ~ALC5632_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5632_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5632_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5632_DAI_I2S_DL_24; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", + __func__, alc5632->sysclk, rate, coeff); + snd_soc_write(codec, ALC5632_DAC_CLK_CTRL1, coeff); + + return 0; +} + +static int alc5632_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5632_MISC_HP_DEPOP_MUTE_L | ALC5632_MISC_HP_DEPOP_MUTE_R; + u16 mute_reg = snd_soc_read(codec, ALC5632_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5632_MISC_CTRL, mute_reg); +} + +#define ALC5632_ADD2_POWER_EN ( ALC5632_PWR_ADD2_VREF ) + +#define ALC5632_ADD3_POWER_EN ( \ + ALC5632_PWR_ADD3_MIC1_BOOST_AD ) + +#define ALC5632_ADD1_POWER_EN \ + ( ALC5632_PWR_ADD1_SPK_AMP_EN \ + | ALC5632_PWR_ADD1_DAC_REF \ + | ALC5632_PWR_ADD1_DAC_L_EN \ + | ALC5632_PWR_ADD1_DAC_R_EN \ + | ALC5632_PWR_ADD1_SOFTGEN_EN \ + | ALC5632_PWR_ADD1_MAIN_I2S_EN | ALC5632_PWR_ADD1_HP_OUT_AMP \ + | ALC5632_PWR_ADD1_HP_OUT_ENH_AMP \ + | ALC5632_PWR_ADD1_MAIN_BIAS ) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_SOFTGEN_EN, + ALC5632_PWR_ADD1_SOFTGEN_EN); + + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD3, ALC5632_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + ALC5632_MISC_HP_DEPOP_MODE2_EN); + + msleep(500); + + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, ALC5632_ADD2_POWER_EN); + + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, ALC5632_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5632_MISC_CTRL, + ALC5632_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5632_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, + ALC5632_PWR_ADD1_MAIN_BIAS); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, + ALC5632_PWR_ADD2_VREF); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD2, 0); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD3, 0); + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define ALC5632_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5632_dai_ops = { + .hw_params = alc5632_pcm_hw_params, + .digital_mute = alc5632_mute, + .set_fmt = alc5632_set_dai_fmt, + .set_sysclk = alc5632_set_dai_sysclk, + .set_pll = alc5632_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5632_dai = { + .name = "alc5632-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5632_FORMATS,}, + + .ops = &alc5632_dai_ops, +}; + +static int alc5632_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5632_resume(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) + snd_soc_write(codec, i, cache[i]); + + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* charge alc5632 caps */ + if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->dapm.bias_level = SND_SOC_BIAS_ON; + alc5632_set_bias_level(codec, codec->dapm.bias_level); + } + + return 0; +} + +#define ALC5632_REC_UNMUTE ( ALC5632_ADC_REC_MIC2 | \ + ALC5632_ADC_REC_LINE_IN | ALC5632_ADC_REC_AUX | \ + ALC5632_ADC_REC_HP | ALC5632_ADC_REC_SPK | \ + ALC5632_ADC_REC_MONOMIX ) + +#define ALC5632_MIC_ROUTE ( ALC5632_MIC_ROUTE_HP | \ + ALC5632_MIC_ROUTE_SPK | ALC5632_MIC_ROUTE_MONOMIX ) + +#define ALC5632_PWR_DEFAULT ( ALC5632_PWR_ADC_STATUS | \ + ALC5632_PWR_DAC_STATUS | ALC5632_PWR_AMIX_STATUS | \ + ALC5632_PWR_VREF_STATUS ) + +#define ALC5632_ADC_REC_GAIN_COMP(x) (int)(( x - ALC5632_ADC_REC_GAIN_BASE ) \ + / ALC5632_ADC_REC_GAIN_STEP) + +#define ALC5632_MIC_BOOST_COMP(x) (int)( x / ALC5632_MIC_BOOST_STEP ) + +#define ALC5632_SPK_OUT_VOL_COMP(x) (int)( x / ALC5632_SPK_OUT_VOL_STEP ) + +static int alc5632_probe(struct snd_soc_codec *codec) +{ + struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + dev_dbg(codec->dev, "ALC5632 Audio Codec Version %s\n", ALC5632_VERSION); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5632->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5632_reset(codec); + + /* power on device */ + alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (alc5632->add_ctrl) { + snd_soc_write(codec, ALC5632_PWR_MANAG_ADD1, + alc5632->add_ctrl); + } + + /* spk amp pwr enable 3A | 0x0400 @ 3A */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD1, + 0, ALC5632_PWR_ADD1_SPK_AMP_EN); + + /* "normal" mode: 0 @ 26 */ + snd_soc_write(codec, ALC5632_PWR_DOWN_CTRL_STATUS, 0); + + /* power on VREF on all analog circuits 0x2000 @ 3C */ + snd_soc_update_bits(codec, ALC5632_PWR_MANAG_ADD2, + 0, ALC5632_PWR_ADD2_VREF); + + /* enable slave mode 0x8000 @ 34 */ + snd_soc_write(codec, ALC5632_DAI_CONTROL, ALC5632_DAI_SDP_SLAVE_MODE); + + switch (alc5632->id) { + case 0x5c: + snd_soc_add_controls(codec, alc5632_vol_snd_controls, + ARRAY_SIZE(alc5632_vol_snd_controls)); + break; + default: + return -EINVAL; + } + + snd_soc_add_controls(codec, alc5632_snd_controls, + ARRAY_SIZE(alc5632_snd_controls)); + + snd_soc_dapm_new_controls(dapm, alc5632_dapm_widgets, + ARRAY_SIZE(alc5632_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + return ret; +} + +/* power down chip */ +static int alc5632_remove(struct snd_soc_codec *codec) +{ + alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5632 = { + .probe = alc5632_probe, + .remove = alc5632_remove, + .suspend = alc5632_suspend, + .resume = alc5632_resume, + .set_bias_level = alc5632_set_bias_level, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = alc5632_reg_defaults, + .reg_cache_size = ARRAY_SIZE(alc5632_reg_defaults), + .volatile_register = alc5632_volatile_register, +}; + +/* + * alc5632 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5632_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5632_platform_data *pdata; + struct alc5632_priv *alc5632; + int ret, vid1, vid2; + + dev_dbg(&client->dev, "in i2c probe...\n"); + + vid1 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_err(&client->dev, "got vid1: %x\n", vid1); + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_word_data(client, ALC5632_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } else { + dev_err(&client->dev, "got vid2: %x\n", vid2); + } + vid2 = (vid2 & 0xff); + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + dev_dbg(&client->dev, "Found codec: ALC5632 (ID: %x)\n", vid2); + + alc5632 = kzalloc(sizeof(struct alc5632_priv), GFP_KERNEL); + if (alc5632 == NULL) + return -ENOMEM; + + pdata = client->dev.platform_data; + if (pdata) { + alc5632->add_ctrl = pdata->add_ctrl; + alc5632->jack_det_ctrl = pdata->jack_det_ctrl; + } + + alc5632->id = vid2; + switch (alc5632->id) { + case 0x5c: + alc5632_dai.name = "alc5632-hifi"; + break; + default: + kfree(alc5632); + return -EINVAL; + } + + i2c_set_clientdata(client, alc5632); + alc5632->control_data = client; + alc5632->control_type = SND_SOC_I2C; + mutex_init(&alc5632->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5632, &alc5632_dai, 1); + if (ret != 0) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(alc5632); + } + + return ret; +} + +static int alc5632_i2c_remove(struct i2c_client *client) +{ + struct alc5632_priv *alc5632 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(alc5632); + return 0; +} + +static const struct i2c_device_id alc5632_i2c_table[] = { + {"alc5632", 0x5c}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5632_i2c_driver = { + .driver = { + .name = "alc5632", + .owner = THIS_MODULE, + }, + .probe = alc5632_i2c_probe, + .remove = __devexit_p(alc5632_i2c_remove), + .id_table = alc5632_i2c_table, +}; + +static int __init alc5632_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5632_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5632_modinit); + +static void __exit alc5632_modexit(void) +{ + i2c_del_driver(&alc5632_i2c_driver); +} +module_exit(alc5632_modexit); + +MODULE_DESCRIPTION("ASoC ALC5632 driver"); +MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h new file mode 100644 index 0000000..b28ae24 --- /dev/null +++ b/sound/soc/codecs/alc5632.h @@ -0,0 +1,233 @@ +/* +* alc5632.h -- ALC5632 ALSA SoC Audio Codec +* +* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> +* +* Authors: Leon Romanovsky <leon(a)leon.nu> +* Andrey Danin <danindrey(a)mail.ru> +* Ilya Petrov <ilya.muromec(a)gmail.com> +* Marc Dietrich <marvin24(a)gmx.de> +* +* Based on alc5623.h by Arnaud Patard +* +* 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 _ALC5632_H +#define _ALC5632_H + +#define ALC5632_RESET 0x00 +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5632_SPK_OUT_VOL 0x02 // spe out vol +#define ALC5632_SPK_OUT_VOL_STEP 1.5 +#define ALC5632_HP_OUT_VOL 0x04 // hp out vol +#define ALC5632_AUX_OUT_VOL 0x06 // aux out vol +#define ALC5632_PHONE_IN_VOL 0x08 // phone in vol +#define ALC5632_LINE_IN_VOL 0x0A // line in vol +#define ALC5632_STEREO_DAC_IN_VOL 0x0C // stereo dac in vol +#define ALC5632_MIC_VOL 0x0E // mic in vol +#define ALC5632_MIC_ROUTING_CTRL 0x10 // stero dac/mic routing +#define ALC5632_MIC_ROUTE_MONOMIX (1 << 0) +#define ALC5632_MIC_ROUTE_SPK (1 << 1) +#define ALC5632_MIC_ROUTE_HP (1 << 2) + +#define ALC5632_ADC_REC_GAIN 0x12 // rec gain +#define ALC5632_ADC_REC_GAIN_RANGE 0x1F1F +#define ALC5632_ADC_REC_GAIN_BASE -16.5 +#define ALC5632_ADC_REC_GAIN_STEP 1.5 + +#define ALC5632_ADC_REC_MIXER 0x14 // mixer control +#define ALC5632_ADC_REC_MIC1 (1 << 6) +#define ALC5632_ADC_REC_MIC2 (1 << 5) +#define ALC5632_ADC_REC_LINE_IN (1 << 4) +#define ALC5632_ADC_REC_AUX (1 << 3) +#define ALC5632_ADC_REC_HP (1 << 2) +#define ALC5632_ADC_REC_SPK (1 << 1) +#define ALC5632_ADC_REC_MONOMIX (1 << 0) + +#define ALC5632_VOICE_DAC_VOL 0x18 // voice dac vol +/* ALC5632_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5632_OUTPUT_MIXER_CTRL 0x1C // out mix ctrl +#define ALC5632_OUTPUT_MIXER_RP (1 << 14) +#define ALC5632_OUTPUT_MIXER_WEEK (1 << 12) +#define ALC5632_OUTPUT_MIXER_HP (1 << 10) +#define ALC5632_OUTPUT_MIXER_AUX_SPK (2 << 6) +#define ALC5632_OUTPUT_MIXER_AUX_HP_LR (1 << 6) +#define ALC5632_OUTPUT_MIXER_HP_R (1 << 8) +#define ALC5632_OUTPUT_MIXER_HP_L (1 << 9) + +#define ALC5632_MIC_CTRL 0x22 // mic phone ctrl +#define ALC5632_MIC_BOOST_BYPASS 0 +#define ALC5632_MIC_BOOST_20DB 1 +#define ALC5632_MIC_BOOST_30DB 2 +#define ALC5632_MIC_BOOST_40DB 3 + +#define ALC5632_DIGI_BOOST_CTRL 0x24 // digi mic / bost ctl +#define ALC5632_MIC_BOOST_RANGE 7 +#define ALC5632_MIC_BOOST_STEP 6 +#define ALC5632_PWR_DOWN_CTRL_STATUS 0x26 +#define ALC5632_PWR_VREF_STATUS (1 << 3) +#define ALC5632_PWR_AMIX_STATUS (1 << 2) +#define ALC5632_PWR_DAC_STATUS (1 << 1) +#define ALC5632_PWR_ADC_STATUS (1 << 0) +#define ALC5632_DAC_FUNC_SELECT 0x2E // stereo/voice DAC / stereo adc func ctrl + +#define ALC5632_DAI_CONTROL 0x34 // Main serial data port ctrl (i2s) + +#define ALC5632_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5632_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5632_DAI_SADLRCK_MODE (1 << 14) +#define ALC5632_DAI_MAIN_I2S_SYSCLK_SEL (1 << 8) // 0:voice, 1:main +#define ALC5632_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +#define ALC5632_DAI_MAIN_I2S_LRCK_INV (1 << 6) // 0:normal, 1:invert +#define ALC5632_DAI_I2S_DL_MASK (3 << 2) +#define ALC5632_DAI_I2S_DL_8 (3 << 2) +#define ALC5632_DAI_I2S_DL_24 (2 << 2) +#define ALC5632_DAI_I2S_DL_20 (1 << 2) +#define ALC5632_DAI_I2S_DL_16 (0 << 2) +#define ALC5632_DAI_I2S_DF_MASK (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_B (3 << 0) +#define ALC5632_DAI_I2S_DF_PCM_A (2 << 0) +#define ALC5632_DAI_I2S_DF_LEFT (1 << 0) +#define ALC5632_DAI_I2S_DF_I2S (0 << 0) + +#define ALC5632_DAI_CONTROL2 0x36 // extend serial data port control (VoDAC_i2c/pcm) + +#define ALC5632_DAI_VOICE_PCM_ENABLE (1 << 15) // 0:gpio func, 1:voice pcm +#define ALC5632_DAI_VOICE_MODE_SEL (1 << 14) // 0:master, 1:slave +#define ALC5632_DAI_HPF_CLK_CTRL (1 << 13) // 0:disable, 1:enable +#define ALC5632_DAI_VOICE_I2S_SYSCLK_SEL (1 << 8) // 0:main, 1:voice +#define ALC5632_DAI_VOICE_VBCLK_SYSCLK_SEL (1 << 7) // 0:normal, 1:invert +#define ALC5632_DAI_VOICE_I2S_LR_INV (1 << 6) // 0:normal, 1:invert +#define ALC5632_DAI_VOICE_DL_MASK (3 << 2) +#define ALC5632_DAI_VOICE_DL_16 (0 << 2) +#define ALC5632_DAI_VOICE_DL_20 (1 << 2) +#define ALC5632_DAI_VOICE_DL_24 (2 << 2) +#define ALC5632_DAI_VOICE_DL_8 (3 << 2) +#define ALC5632_DAI_VOICE_DF_MASK (3 << 0) +#define ALC5632_DAI_VOICE_DF_I2S (0 << 0) +#define ALC5632_DAI_VOICE_DF_LEFT (1 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_A (2 << 0) +#define ALC5632_DAI_VOICE_DF_PCM_B (3 << 0) + +#define ALC5632_PWR_MANAG_ADD1 0x3A +#define ALC5632_PWR_ADD1_DAC_L_EN (1 << 15) +#define ALC5632_PWR_ADD1_DAC_R_EN (1 << 14) +#define ALC5632_PWR_ADD1_ZERO_CROSS (1 << 13) +#define ALC5632_PWR_ADD1_MAIN_I2S_EN (1 << 11) +#define ALC5632_PWR_ADD1_SPK_AMP_EN (1 << 10) +#define ALC5632_PWR_ADD1_HP_OUT_AMP (1 << 9) +#define ALC5632_PWR_ADD1_HP_OUT_ENH_AMP (1 << 8) +#define ALC5632_PWR_ADD1_VOICE_DAC_MIX (1 << 7) +#define ALC5632_PWR_ADD1_SOFTGEN_EN (1 << 6) +#define ALC5632_PWR_ADD1_MIC1_SHORT_CURR (1 << 5) +#define ALC5632_PWR_ADD1_MIC2_SHORT_CURR (1 << 4) +#define ALC5632_PWR_ADD1_MIC1_EN (1 << 3) +#define ALC5632_PWR_ADD1_MIC2_EN (1 << 2) +#define ALC5632_PWR_ADD1_MAIN_BIAS (1 << 1) +#define ALC5632_PWR_ADD1_DAC_REF (1 << 0) + +#define ALC5632_PWR_MANAG_ADD2 0x3C +#define ALC5632_PWR_ADD2_PLL1 (1 << 15) +#define ALC5632_PWR_ADD2_PLL2 (1 << 14) +#define ALC5632_PWR_ADD2_VREF (1 << 13) +#define ALC5632_PWR_ADD2_OVT_DET (1 << 12) +#define ALC5632_PWR_ADD2_VOICE_DAC (1 << 10) +#define ALC5632_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5632_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5632_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5632_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5632_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5632_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5632_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5632_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5632_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5632_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5632_PWR_MANAG_ADD3 0x3E +#define ALC5632_PWR_ADD3_AUXOUT_VOL (1 << 14) +#define ALC5632_PWR_ADD3_SPK_L_OUT (1 << 13) +#define ALC5632_PWR_ADD3_SPK_R_OUT (1 << 12) +#define ALC5632_PWR_ADD3_HP_L_OUT_VOL (1 << 11) +#define ALC5632_PWR_ADD3_HP_R_OUT_VOL (1 << 10) +#define ALC5632_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5632_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5632_PWR_ADD3_AUXIN_VOL (1 << 5) +#define ALC5632_PWR_ADD3_AUXIN_MIX (1 << 4) +#define ALC5632_PWR_ADD3_MIC1_VOL (1 << 3) +#define ALC5632_PWR_ADD3_MIC2_VOL (1 << 2) +#define ALC5632_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5632_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5632_GPCR1 0x40 +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_PLL1 (1 << 15) +#define ALC5632_GPCR1_CLK_SYS_SRC_SEL_MCLK (0 << 15) +#define ALC5632_GPCR1_DAC_HI_FLT_EN (1 << 10) +#define ALC5632_GPCR1_SPK_AMP_CTRL (7 << 1) +#define ALC5632_GPCR1_VDD_100 (5 << 1) +#define ALC5632_GPCR1_VDD_125 (4 << 1) +#define ALC5632_GPCR1_VDD_150 (3 << 1) +#define ALC5632_GPCR1_VDD_175 (2 << 1) +#define ALC5632_GPCR1_VDD_200 (1 << 1) +#define ALC5632_GPCR1_VDD_225 (0 << 1) + +#define ALC5632_GPCR2 0x42 +#define ALC5632_GPCR2_PLL1_SOUR_SEL (3 << 12) +#define ALC5632_PLL_FR_MCLK (0 << 12) +#define ALC5632_PLL_FR_BCLK (2 << 12) +#define ALC5632_PLL_FR_VBCLK (3 << 12) +#define ALC5632_GPCR2_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5632_PLL1_CTRL 0x44 +#define ALC5632_PLL1_CTRL_N_VAL(n) (((n) & 0x0f) << 8) +#define ALC5632_PLL1_M_BYPASS (1 << 7) +#define ALC5632_PLL1_CTRL_K_VAL(k) (((k) & 0x07) << 4) +#define ALC5632_PLL1_CTRL_M_VAL(m) (((m) & 0x0f) << 0) + +#define ALC5632_PLL2_CTRL 0x46 +#define ALC5632_PLL2_EN (1 << 15) +#define ALC5632_PLL2_RATIO (0 << 15) + +#define ALC5632_GPIO_PIN_CONFIG 0x4C +#define ALC5632_GPIO_PIN_POLARITY 0x4E +#define ALC5632_GPIO_PIN_STICKY 0x50 +#define ALC5632_GPIO_PIN_WAKEUP 0x52 +#define ALC5632_GPIO_PIN_STATUS 0x54 +#define ALC5632_GPIO_PIN_SHARING 0x56 +#define ALC5632_OVER_CURR_STATUS 0x58 +#define ALC5632_SOFTVOL_CTRL 0x5A +#define ALC5632_GPIO_OUPUT_PIN_CTRL 0x5C + +#define ALC5632_MISC_CTRL 0x5E +#define ALC5632_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5632_MISC_AVC_TRGT_SEL (3 << 12) +#define ALC5632_MISC_AVC_TRGT_RIGHT (1 << 12) +#define ALC5632_MISC_AVC_TRGT_LEFT (2 << 12) +#define ALC5632_MISC_AVC_TRGT_BOTH (3 << 12) +#define ALC5632_MISC_HP_DEPOP_MODE1_EN (1 << 9) +#define ALC5632_MISC_HP_DEPOP_MODE2_EN (1 << 8) +#define ALC5632_MISC_HP_DEPOP_MUTE_L (1 << 7) +#define ALC5632_MISC_HP_DEPOP_MUTE_R (1 << 6) +#define ALC5632_MISC_HP_DEPOP_MUTE (1 << 5) +#define ALC5632_MISC_GPIO_WAKEUP_CTRL (1 << 1) +#define ALC5632_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5632_DAC_CLK_CTRL1 0x60 +#define ALC5632_DAC_CLK_CTRL2 0x62 +#define ALC5632_DAC_CLK_CTRL2_DIV1_2 (1 << 0) +#define ALC5632_VOICE_DAC_PCM_CLK_CTRL1 0x64 +#define ALC5632_PSEUDO_SPATIAL_CTRL 0x68 +#define ALC5632_HID_CTRL_INDEX 0x6A +#define ALC5632_HID_CTRL_DATA 0x6C +#define ALC5632_EQ_CTRL 0x6E + +/* undocumented */ +#define ALC5632_VENDOR_ID1 0x7C +#define ALC5632_VENDOR_ID2 0x7E + +#endif -- 1.7.3.4

1 0

Re: [alsa-devel] ALSA PulseAudio plugin: snd_pcm_rewindable() returns >0, but snd_pcm_rewind() is broken
by Nikolay Nikolov 25 Oct '11

25 Oct '11

On 10/24/2011 06:34 PM, Pierre-Louis Bossart wrote: > You want to avoid rewinding completely. Your audio hardware might have > prefetched data with the DMA subsystem. Rewinding completely might result in > an inconsistent configuration and possibly underflows. If you look at the > PulseAudio code, we've introduced some thresholds beyond which we don't > rewind (128 bytes or 1ms off the top of my head). > You might argue that snd_pcm_rewindable is broken, but it's somewhat > difficult to fix as the amount of prefetched data isn't modeled in the > driver and it's very much hardware-specific. Using a less aggressive > approach works fine on most hardware. > -Pierre Yes, I figured this out while experimenting with real hw alsa devices and that's why I actually rewind snd_pcm_rewindable() minus some threshold, that can be configured. However with the alsa pulseaudio plugin, rewinding seems to be broken completely, even with a very large threshold.

2 1

[alsa-devel] ALSA PulseAudio plugin: snd_pcm_rewindable() returns >0, but snd_pcm_rewind() is broken
by Nikolay Nikolov 24 Oct '11

24 Oct '11

Hi all, I submitted a bug report for this here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5464 I got a comment, which said I should post this on the mailing list, so I'm doing this now: I'm writing a program that tries to use the snd_pcm_rewind() function if it's available and I'm having serious trouble with the alsa-pulseaudio plugin. I'm using the max buffer size, which is 1048576 frames (4MB buffer, 16-bit stereo). The buffer is almost full when I try to rewind. I first call snd_pcm_rewindable() and it returns a greater than zero value. Then I call snd_pcm_rewind() with this value and it returns success, i.e. a positive value of frames actually rewound. I then call snd_pcm_writei with the number of frames rewound and I get a weird pause (several seconds) in the audio on the next poll. Audio then continues from the middle; the beginning of the audio that was rewritten immediately after rewind() has been dropped. Also, pulseaudio reports several times the following error: protocol-native.c: Failed to push data into queue After looking at the alsa-lib and alsa-plugins sources, it seems that snd_pcm_rewind() for ioplug devices only updates an internal pointer and doesn't notify pulseaudio, so on the next write we just try to push too much data to the pulseaudio stream. After googling a bit, I found this email from 17 Feb 2010 from Lennart Poettering, which says that snd_pcm_rewind() is broken for the alsa-pulseaudio plugin. http://ns.spinics.net/lists/alsa-devel/msg31536.html Also, this mail: http://lists.freedesktop.org/archives/pulseaudio-bugs/2010-July/004051.html says that snd_pcm_rewindable() should always return 0 frames on ioplug devices, and if that's not the case, then it's a bug in alsa. So, if I'm getting this right, either snd_pcm_rewindable() should be fixed to return 0, or snd_pcm_rewind() should be fixed to work properly with the alsa-pulseaudio plugin. Thanks, Nikolay

2 1

[alsa-devel] dmix - unable to open slave
by Nenad Sljivic 24 Oct '11

24 Oct '11

Hi, I posted the same questions on alsa-user group, but no one could help me, so I might have more luck here. I am trying to enable a software mixer on an MIPS based embedded target board using alsa-lib v1.0.23. When I try to play two audio files using these commands: #aplay -vDplug:dmix song1.wav & #aplay -vDplug:dmix song2.wav mixing works fine. For some strange reason when I try the same thing using the "nenad_dmix" device I am getting the following error: ALSA lib pcm_dmix.c:1068:(snd_pcm_dmix_open) unable to open slave aplay: main:654: audio open error: Invalid argument The error occurs while trying to open device for the second time (song2.wav playback). At the same time, song1.wav plays fine. The content of my .asoundrc is given bellow: pcm.nenad_dmix { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0666 slave { pcm "hw:0,0" period_time 0 period_size 3840 buffer_size 15360 rate 48000 } } I narrowed down the problem using the strace tool and it showed me this: open("/dev/snd/pcmC0D0p", O_RDWR|O_APPEND|O_NONBLOCK|O_CLOEXEC) = -1 EINVAL (Invalid argument) What can be the difference between using the plug:dmix and some virtual device like nenad_dmix in this case ?? Could some of you help me to resolve this issue? Thanks, Nenad

1 0

Re: [alsa-devel] Is locking problem in snd_intel8x0_pcm_pointer() ?
by Takashi Iwai 24 Oct '11

24 Oct '11

At Mon, 24 Oct 2011 15:48:44 +0400, Konstantin Ozerkov wrote: > > Seems that locking is broken: snd_intel8x0_pcm_pointer() uses spin_lock()/spin_unlock() semantic > for lock which is shared in ISR ( snd_intel8x0_update). As result we got possible deadlock inside ISR > on UP system. And needed special kludge inside snd_intel8x0_pcm_pointer() to prevent moving > backward (really this is race with snd_intel8x0_update). Hm, could you elaborate how can it a problem? snd_intel8x0_update() releases the lock before calling snd_pcm_period_update() that calls the pointer callback. Takashi

1 0

Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Mark Brown 24 Oct '11

24 Oct '11

On Mon, Oct 24, 2011 at 09:12:42PM +0900, Tomoya MORINAGA wrote: > 1. PCI interface function. > Any current ASoC drivers don't have PCI interface function. > So I don't know where the function should be in machine driver or > platform driver. It depends on what the driver is for. Probably you want a driver which is some combination of machine driver and the various drivers that are normally part of the SoC - whatever roles in the system are filled by this hardware the driver ought to register subsystem drivers for those roles. > 2. Register Access > Can platform driver access register ? > According to the soc document, platform driver must not access hardware, > however, some drivers looks accessing their hardware. What makes you say this? A driver that can't access hardware would be rather useless...

1 0

[alsa-devel] [PATCH 1/3] ASoC: wm8940: Fix setting PLL Output clock division ratio
by Axel Lin 24 Oct '11

24 Oct '11

According to the datasheet: The PLL Output clock division ratio is controlled by BIT[5:4] of WM8940_GPIO register(08h). Current code read/write the WM8940_ADDCNTRL(07h) register which is wrong. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8940.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index a4abfdf..3cc3bce 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -627,8 +627,8 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 5)); break; case WM8940_OPCLKDIV: - reg = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFCF; - ret = snd_soc_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); + reg = snd_soc_read(codec, WM8940_GPIO) & 0xFFCF; + ret = snd_soc_write(codec, WM8940_GPIO, reg | (div << 4)); break; } return ret; -- 1.7.5.4

3 6

[alsa-devel] [PATCH] ASoC: wm8996: Avoid a redundant i2c_get_clientdata call in wm8996_i2c_remove
by Axel Lin 24 Oct '11

24 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 645c980..32324c9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3144,7 +3144,7 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); if (wm8996->pdata.ldo_ena > 0) gpio_free(wm8996->pdata.ldo_ena); - kfree(i2c_get_clientdata(client)); + kfree(wm8996); return 0; } -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: wm8400: Fix setting Fout clock divider for FLL Control 4
by Axel Lin 24 Oct '11

24 Oct '11

What we want here is to clear the WM8400_FLL_OUTDIV_MASK bits then OR with factors.outdiv. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8400.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index dc13be2..f29bc26 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1059,7 +1059,7 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); - reg &= WM8400_FLL_OUTDIV_MASK; + reg &= ~WM8400_FLL_OUTDIV_MASK; reg |= factors.outdiv; wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ALSA: HDA: Add new revision for ALC662
by David Henningsson 24 Oct '11

24 Oct '11

We found a new chip, with Vendor Id: 0x10ec0662 and Revision Id: 0x100300. It seems to work well with patch_alc662, but I don't know if patch_alc882 had been better, and actually, I don't know if there's much of a difference between the two patch functions these days anyway. I'm cc:ing Kailang in case there's more to it (codec renaming, processing coefficients etc) that we don't know of. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] Use of _hint() functions and older machines
by Randell Jesup 24 Oct '11

24 Oct '11

[ I initially posted this to the -users list, but it may be more appropriate here ] At Mozilla, we're in the process of adding support for WebRTC (http://webrtc.org/), which is being standardized by the IETF (their part is 'rtcweb'), and the W3C. This adds real-time audio and video (and data) communication to browsers, peer-to-peer over encrypted channels. We have a sound library that can load either Pulse or Alsa. However, for Alsa, it wants to look at snd_device_name_hint() and also _get_hint() and _free_hint(). It lazy-binds to libasound, so it will dlopen() it and then dlsym() all the symbols it uses; if any fail it unloads the lib and says it's not there. It uses the hint functions to build a device list, for example for presenting to the user. I have two problems: 1) Firefox is build on machines configured with I believe Centos5, and I'm told the machines run Alsa 1.0.12, while the hints() functions were added in 1.0.14 (released June 2007). Right now I can't build release or 'try' builds on the build servers because of this. 2) We'd like to run on older machines if possible, and official release builds are made on those servers. On older machines, _hint() aren't available, so even if I make them optional to dlsym-loading, I would need some other method to get the information I assume using older, now-deprecated-or-gone interfaces. Any suggestions? Tips? See http://code.google.com/p/webrtc/source/browse/trunk/src/modules/audio_devic… for how they're used currently. Thanks! -- Randell Jesup randell-ietf(a)jesup.org

4 6

[alsa-devel] Register cache different from the real register values
by Leon Romanovsky 24 Oct '11

24 Oct '11

Hi, I need your suggestions, I'm trying to add new codec to use the asoc subsystem, all read/write operations done via standard snd_soc_* functions with cache enabled, but register cache is different from the real register values. Can you point me to what can I check? Our code is based on 2.6.38 release. Thanks -- Leon Romanovsky | Independent Linux Consultant www.leon.nu | leon(a)leon.nu

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: core: Ignoring pmdown_time for playback stream power down
by Peter Ujfalusi 24 Oct '11

24 Oct '11

Hello Mark, Changes since v1 (RFC): - move the flag to per codec (snd_soc_codec) - twl6040 to request to ignore the pmdown_time Add support in core to ignore the pmdown_time when stopping the playback stream. This will inline the DAPM stream stop event, so it is going to be executed without delay at stream close time (for playback stream). Backround: with the McPDM protocol we are sendning not only the pure audio stream, but OMAP McPDM also transmits additional information (for example offset cancellation). If McPDM is stopped prior to the DAC this information will be not sent to the codec, which can result noise rendered by the twl6040 codec. Regards, Peter --- Peter Ujfalusi (2): ASoC: core: Add flag to ignore pmdown_time at pcm_close ASoC: twl6040: Request core to inline the DAPM sequence include/sound/soc.h | 1 + sound/soc/codecs/twl6040.c | 1 + sound/soc/soc-pcm.c | 15 +++++++++++---- 3 files changed, 13 insertions(+), 4 deletions(-) -- 1.7.7

5 17

[alsa-devel] ASoC updates for 3.2
by Mark Brown 22 Oct '11

22 Oct '11

The following changes since commit 24441e17feb57668e4c5013750cef549bf3c4861: ASoC: ak4641: Use SND_SOC_DAPM_DAC for Voice Playback stream widget (2011-10-19 18:01:43 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Ashish Chavan (2): ASoC: da7210: Add support for DAPM ASoC: da7210: Add support for line out and DAC Axel Lin (10): ASoC: wm9081: Fix setting soft VMID ramp enable with VMID 2*240k ASoC: Fix reg_word_size for ak4104 ASoC: ak4671: Use snd_soc_update_bits for read-modify-write ASoC: rt5631: Remove unused codec field from struct rt5631_priv ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture ASoC: Set sgtl5000->ldo in ldo_regulator_register ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2 ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls Julia Lawall (1): ASoC: keep pointer to resource so it can be freed sound/soc/au1x/ac97c.c | 33 +++--- sound/soc/au1x/i2sc.c | 33 +++--- sound/soc/au1x/psc-ac97.c | 25 +++-- sound/soc/au1x/psc-i2s.c | 25 +++-- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4671.c | 18 +-- sound/soc/codecs/da7210.c | 241 ++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt5631.c | 2 - sound/soc/codecs/sgtl5000.c | 20 ++-- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8996.c | 6 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/mxs/mxs-saif.c | 19 ++-- 13 files changed, 312 insertions(+), 116 deletions(-)

2 1

[alsa-devel] [PATCH v2] ASoC: max98095: Convert codec->hw_write to snd_soc_write
by Axel Lin 22 Oct '11

22 Oct '11

codec->hw_write is broken now, convert codec->hw_write to snd_soc_write. The hardware has 2 banks of registers sharing a section in I2C register space. The 1st bank is the primary one and is cached. The 2nd bank is for loading coefficients only and they do not need cache. These coefficients registers are therefore direct writes. Thus we set cache_bypass flag to deal with this before calling snd_soc_write. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi, I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/max98095.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 6982f74..26d7b08 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -617,14 +617,13 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - u8 data[2]; + int ret; - data[0] = reg; - data[1] = value; - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; + codec->cache_bypass = 1; + ret = snd_soc_write(codec, reg, value); + codec->cache_bypass = 0; + + return ret ? -EIO : 0; } /* -- 1.7.4.1

2 1

[alsa-devel] [PATCH] sound/soc: keep pointer to resource so it can be freed
by Julia Lawall 22 Oct '11

22 Oct '11

From: Julia Lawall <julia(a)diku.dk> Add a new variable for storing resources accessed subsequent to the one accessed using request_mem_region, so the one accessed using request_mem_region can be released if needed. The resource variable names are also changed to be more descriptive. This code is also missing some calls to iounmap. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ expression E, E1; identifier f; statement S1,S2,S3; @@ if (E == NULL) { ... when != if (E == NULL || ...) S1 else S2 when != E = E1 *E->f ... when any return ...; } else S3 // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/au1x/ac97c.c | 33 ++++++++++++++++++--------------- sound/soc/au1x/i2sc.c | 33 ++++++++++++++++++--------------- sound/soc/au1x/psc-ac97.c | 25 +++++++++++++------------ sound/soc/au1x/psc-i2s.c | 25 +++++++++++++------------ sound/soc/mxs/mxs-saif.c | 19 ++++++++++--------- 5 files changed, 72 insertions(+), 63 deletions(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 13802ff..726bd65 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -226,7 +226,7 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = { static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); @@ -235,29 +235,30 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) mutex_init(&ctx->lock); - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* switch it on */ WR(ctx, AC97_ENABLE, EN_D | EN_CE); @@ -270,13 +271,15 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); if (ret) - goto out1; + goto out2; ac97c_workdata = ctx; return 0; +out2: + iounmap(ctx->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(ctx); return ret; diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 19e0d2a..6bcf48f 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -228,47 +228,50 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - ctx->mmio = ioremap_nocache(r->start, resource_size(r)); + ctx->mmio = ioremap_nocache(iores->start, resource_size(iores)); if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; platform_set_drvdata(pdev, ctx); ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); if (ret) - goto out1; + goto out2; return 0; +out2: + iounmap(ctx->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(ctx); return ret; diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 172eefd..cab74ab 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -364,7 +364,7 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; struct au1xpsc_audio_data *wd; @@ -374,29 +374,30 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) mutex_init(&wd->lock); - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | @@ -428,7 +429,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) out2: iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 7c5ae92..d792bc0 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -290,7 +290,7 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { - struct resource *r; + struct resource *iores, *dmares; unsigned long sel; int ret; struct au1xpsc_audio_data *wd; @@ -299,29 +299,30 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd) return -ENOMEM; - r = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!r) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; goto out0; } ret = -EBUSY; - if (!request_mem_region(r->start, resource_size(r), pdev->name)) + if (!request_mem_region(iores->start, resource_size(iores), + pdev->name)) goto out0; - wd->mmio = ioremap(r->start, resource_size(r)); + wd->mmio = ioremap(iores->start, resource_size(iores)); if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmares) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -355,7 +356,7 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) out2: iounmap(wd->mmio); out1: - release_mem_region(r->start, resource_size(r)); + release_mem_region(iores->start, resource_size(iores)); out0: kfree(wd); return ret; diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 401944c..76dc74d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -617,7 +617,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { - struct resource *res; + struct resource *iores, *dmares; struct mxs_saif *saif; struct mxs_saif_platform_data *pdata; int ret = 0; @@ -655,35 +655,36 @@ static int mxs_saif_probe(struct platform_device *pdev) goto failed_clk; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { + iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!iores) { ret = -ENODEV; dev_err(&pdev->dev, "failed to get io resource: %d\n", ret); goto failed_get_resource; } - if (!request_mem_region(res->start, resource_size(res), "mxs-saif")) { + if (!request_mem_region(iores->start, resource_size(iores), + "mxs-saif")) { dev_err(&pdev->dev, "request_mem_region failed\n"); ret = -EBUSY; goto failed_get_resource; } - saif->base = ioremap(res->start, resource_size(res)); + saif->base = ioremap(iores->start, resource_size(iores)); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; goto failed_ioremap; } - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { + dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmares) { ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); goto failed_ioremap; } - saif->dma_param.chan_num = res->start; + saif->dma_param.chan_num = dmares->start; saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) { @@ -742,7 +743,7 @@ failed_get_irq2: failed_get_irq1: iounmap(saif->base); failed_ioremap: - release_mem_region(res->start, resource_size(res)); + release_mem_region(iores->start, resource_size(iores)); failed_get_resource: clk_put(saif->clk); failed_clk:

2 1

[alsa-devel] [PATCH] ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
by Axel Lin 22 Oct '11

22 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1d7b174..645c980 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1920,7 +1920,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, lrclk_reg, WM8996_AIF1RX_RATE_MASK, lrclk); snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_2, - WM8996_DSP1_DIV_SHIFT << dsp_shift, dsp); + WM8996_DSP1_DIV_MASK << dsp_shift, dsp); return 0; } -- 1.7.5.4

3 2

[alsa-devel] [PATCH v8 2/3] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 22 Oct '11

22 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for OUT1 and OUT2 (Moved here from DAPM patch) - Added comment explaining rational behind usage of STANDBY bits Changes since v1: - Removed explicit setting of default gains - Renamed "DAC Playback" to "Digital Playback" --- sound/soc/codecs/da7210.c | 96 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 96 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 7a4b952..0ebcbd5 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -54,6 +54,9 @@ #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -186,6 +189,17 @@ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ #define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -206,8 +220,23 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -306,6 +335,14 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), @@ -402,6 +439,12 @@ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Side */ @@ -443,16 +486,26 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { &da7210_dapm_outmixr_controls[0], ARRAY_SIZE(da7210_dapm_outmixr_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + /* Output PGAs */ SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), }; /* DAPM audio route definition */ @@ -478,14 +531,26 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + {"Headphone Left", NULL, "OUTPGA Left Enable"}, {"HPL", NULL, "Headphone Left"}, {"Headphone Right", NULL, "OUTPGA Right Enable"}, {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, }; /* Codec private data */ @@ -791,6 +856,37 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable ramp mode for DAC gain update */ snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* + * For DA7210 codec, there are two ways to enable/disable analog IOs + * and ADC/DAC, + * (1) Using "Enable Bit" of register associated with that IO + * (or ADC/DAC) + * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg + * + * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register + * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) + * + * Out of these two methods, the one using STANDBY bits is preferred + * way to enable/disable individual blocks. This is because STANDBY + * registers are part of system controller which allows system power + * up/down in a controlled, pop-free manner. Also, as per application + * note of DA7210, STANDBY register bits are only effective if a + * particular IO (or ADC/DAC) is already enabled using enable/disable + * register bits. Keeping these things in mind, current DAPM + * implementation manipulates only STANDBY bits. + * + * Overall implementation can be outlined as below, + * + * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() + * - "STANDBY bit" is controlled by DAPM + */ + + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

3 2

[alsa-devel] [PATCH v8 1/3] ASoC: da7210: Add support for DAPM
by Ashish Chavan 22 Oct '11

22 Oct '11

This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Re-factored the patch to only include DAPM for existing IOs. (i.e. removed DAPM support for AUX, line outs etc. This will be added via other patches which add actual support for those IOs) Changes since v3: - Added missing routes for In-mixer left and right - Added comment explaining rational behind use of STANDBY registers Changes since v1: - used supply widget for mic bias - removed mic bias hookup - assigned dapm widgets and routes in card struct - removed explicit setting of default mixer settings - removed explicit setting of ios to standby mode --- sound/soc/codecs/da7210.c | 145 +++++++++++++++++++++++++++++++++++++-------- 1 files changed, 121 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e9ee6a4..7a4b952 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -29,6 +29,8 @@ #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_AUX1_L 0x09 @@ -372,6 +374,120 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; +/* + * DAPM Controls + * + * Current DAPM implementation covers almost all codec components e.g. IOs, + * mixers, PGAs,ADC and DAC. + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, +}; + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -411,29 +527,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); - } - - return 0; -} /* * Set PCM DAI word length. @@ -603,7 +696,6 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, .digital_mute = da7210_mute, @@ -742,6 +834,11 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), + + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
by Axel Lin 22 Oct '11

22 Oct '11

Ensure all mask bits are clear before setting new value. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, I'd appreciate if someone can test this patch. Thanks, Axel sound/soc/codecs/sgtl5000.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 32b5bbd..d15695d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -723,7 +723,9 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, + SGTL5000_I2S_DLEN_MASK | SGTL5000_I2S_SCLKFREQ_MASK, + i2s_ctl); return 0; } @@ -1146,8 +1148,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - vag << SGTL5000_ANA_GND_SHIFT, - vag << SGTL5000_ANA_GND_SHIFT); + SGTL5000_ANA_GND_MASK, vag << SGTL5000_ANA_GND_SHIFT); /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ vag = vddio / 2; @@ -1161,9 +1162,8 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) SGTL5000_LINE_OUT_GND_STP; snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - vag << SGTL5000_LINE_OUT_GND_SHIFT | - SGTL5000_LINE_OUT_CURRENT_360u << - SGTL5000_LINE_OUT_CURRENT_SHIFT, + SGTL5000_LINE_OUT_CURRENT_MASK | + SGTL5000_LINE_OUT_GND_MASK, vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT); -- 1.7.5.4

4 3

Re: [alsa-devel] [PATCH v5] ASoC:Add support for cs42l73 codec
by Mark Brown 21 Oct '11

21 Oct '11

On Thu, Oct 20, 2011 at 09:31:50PM +0300, Georgi Vlaev wrote: > Hello, Don't top post. > The codec has 2 basic groups of power control bits: You're describing some features that don't sound at all unusual here... I'm not sure what point you're trying to make here?

3 5

[alsa-devel] [PATCH v7 2/3] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 21 Oct '11

21 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed explicit setting of default gains - Renamed "DAC Playback" to "Digital Playback" --- sound/soc/codecs/da7210.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 42 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 2187db9..eaec60a 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -54,6 +54,8 @@ #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F #define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 @@ -187,6 +189,17 @@ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ #define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + #define DA7210_VERSION "0.0.1" /* @@ -207,8 +220,23 @@ static const unsigned int hp_out_tlv[] = { 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), }; +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -307,6 +335,14 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), @@ -886,6 +922,12 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable ramp mode for DAC gain update */ snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Enable Line out amplifiers */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | + DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

3 8

[alsa-devel] [PATCH v7 1/3] ASoC: da7210: Add support for DAPM
by Ashish Chavan 21 Oct '11

21 Oct '11

This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v3: - Added missing routes for In-mixer left and right - Added comment explaining rational behind use of STANDBY registers Changes since v1: - used supply widget for mic bias - removed mic bias hookup - assigned dapm widgets and routes in card struct - removed explicit setting of default mixer settings - removed explicit setting of ios to standby mode --- sound/soc/codecs/da7210.c | 240 ++++++++++++++++++++++++++++++++++++++++----- 1 files changed, 216 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index f0a1999..2187db9 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -29,6 +29,8 @@ #define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_AUX1_L 0x09 @@ -52,6 +54,7 @@ #define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -372,6 +375,214 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; +/* + * DAPM Controls + * + * Current DAPM implementation covers almost all codec components e.g. IOs, + * mixers, PGAs,ADC and DAC. + * + * For DA7210 codec, there are two ways to enable/disable analog IOs and + * ADC/DAC, + * (1) Using "Enable Bit" of register associated with that IO(or ADC/DAC) + * e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg + * + * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register + * e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5) reg + * + * Out of these two methods, the one using STANDBY bits is preferred way to + * enable/disable individual blocks. This is because STANDBY registers are part + * of system controller which allows system power up/down in a controlled, + * pop-free manner. Also, as per application note of DA7210, STANDBY register + * bits are only effective if a particular IO (or ADC/DAC) is already enabled + * using enable/disable register bits. Keeping these things in mind, current + * DAPM implementation manipulates only STANDBY bits. + * + * Overall implementation can be outlined as below, + * + * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe() function + * - "STANDBY bit" is controlled by DAPM + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, + + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, +}; + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -411,29 +622,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); - } - - return 0; -} /* * Set PCM DAI word length. @@ -603,7 +791,6 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, .digital_mute = da7210_mute, @@ -742,6 +929,11 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), + + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.1

3 7

[alsa-devel] [PATCH] ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
by Axel Lin 21 Oct '11

21 Oct '11

We have defined SGTL5000_LINREG_VDDD_MASK in sgtl5000.h, use it instead of hardcoded (0x1 << 4) - 1 for the mask. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 8395002..32b5bbd 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -754,7 +754,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev) /* set voltage to register */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, reg); + SGTL5000_LINREG_VDDD_MASK, reg); snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_LINEREG_D_POWERUP, @@ -780,7 +780,7 @@ static int ldo_regulator_disable(struct regulator_dev *dev) /* clear voltage info */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, 0); + SGTL5000_LINREG_VDDD_MASK, 0); ldo->enabled = 0; @@ -1115,7 +1115,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* set voltage to register */ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, 0x8); + SGTL5000_LINREG_VDDD_MASK, 0x8); /* * if vddd linear reg has been enabled, -- 1.7.5.4

4 3

[alsa-devel] [PATCH] ASoC: Set sgtl5000->ldo in ldo_regulator_register
by Axel Lin 21 Oct '11

21 Oct '11

Otherwise calling ldo_regulator_remove() does not unregister regulator and free memories. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3637a62..8395002 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -806,6 +806,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, int voltage) { struct ldo_regulator *ldo; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); @@ -840,6 +841,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, return ret; } + sgtl5000->ldo = ldo; return 0; } -- 1.7.5.4

4 3

[alsa-devel] [PATCH 1/2] ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
by Axel Lin 21 Oct '11

21 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 68e769e..03621e2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1419,7 +1419,7 @@ SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux), SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux), SND_SOC_DAPM_AIF_IN("AIF3DACDAT", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0), -- 1.7.5.4

3 3

[alsa-devel] [PATCH v7 3/3] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 21 Oct '11

21 Oct '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index eaec60a..2f38b39 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -928,6 +954,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 2

Re: [alsa-devel] [PATCH 1/1] intel8x0: Improve performance in virtual environment
by Takashi Iwai 20 Oct '11

20 Oct '11

At Thu, 20 Oct 2011 21:32:38 +0400, Konstantin Ozerkov wrote: > > Thank you for fast response and comments. > > I will rewrite this patch according you suggestions: > * add VM environment detection logic (PD only, I don't have clear vision how to > that for VirtualBox/QEMU) and module parameter (for manual control) > * make separated VM-optimized variant of snd_intel8x0_pcm_pointer() Sounds reasonable. > Initially this patch is result of research and profiling audio emulation quality in Parallels Desktop. > > On 20.10.11 19:21, Takashi Iwai wrote: > > At Thu, 20 Oct 2011 19:02:50 +0400, > > Konstantin Ozerkov wrote: > >> This patch intended to improve performance in virtualized environments > >> like Parallels Desktop or VirtualBox/QEMU (virtual ICH/AC97 audio). > >> > >> I/O access is very time-expensive operation in virtual world: VCPU > >> can be rescheduled and in the worst case we got more than 10ms delay on > >> each I/O access. > >> > >> In the original code normal loop exit rule > >> (old_civ == current_civ&& old_picb == current_picb) was never satisfied, > >> because old_picb was never the same as current_picb due delay inspired by > >> reading current_civ. As a result loop ended by timeout and we got 10x more > >> I/O operations. > > You mean in the virtual environment, right? On the real machines, > > this condition can be satisfied normally because the read is much > > faster than the sample rate. This check is needed to avoid the broken > > behavior around the boundary. > > > >> To prevent glitch on buffer swapping the rule (old_civ == current_civ) is > >> enough. > > Not for real (thus less logical) machines, unfortunately. Both checks > > were needed in the real tests. > Can you say which hardware glitches this way and and which conditions? > My simplified rule works well in many Windows AC97 drivers which I debug/analyze. Some hardware showed the unstable CIV and PICB values when it goes crossing the boundary. Both registers should be updated simultaneously, but in reality, you may hit at a bad timing where only one of them is updated. And the order of updates of CIV/PICB isn't defined. Actually, in the earlier version, we tested only CIV, but it turned out that it's didn't suffice. So, we had to check both values. thanks, Takashi

1 0

[alsa-devel] ASoC updates for 3.2
by Mark Brown 20 Oct '11

20 Oct '11

The following changes since commit b29a33a211a855e8e52c095af13e1d99ed60d07c: ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X (2011-10-12 17:09:15 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Ashish Chavan (5): ASoC: da7210: bugfix for head phone volume control ASoC: da7210: Add support for ADC & DAC equalizers ASoC: da7210: Add support for High pass and Voice filters for ADC and DAC ASoC: da7210: Add support for mute and zero cross controls ASoC: da7210: Add support for ALC and Noise suppression Axel Lin (27): ASoC: pxa: Remove redundant snd_soc_dapm_sync() calls from machine drivers ASoC: sta32x: Set reg_cache_default to sta32x_regs ASoC: sta32x: Write the register default value to cache for reserved registers ASoC: ak4535: fixup cache register table ASoC: tlv320aic32x4: Use snd_soc_update_bits for read-modify-write ASoC: tlv320aic23: convert to soc-cache ASoC: alc5623: Convert codec->hw_read to snd_soc_read ASoC: tlv320aic3x: Convert codec->hw_read to snd_soc_read ASoC: wm8961: Convert codec->hw_read to snd_soc_read ASoC: wm8991: Fix wrong bit setting for WM8991_POWER_MANAGEMENT_2 ASoC: wm8991: Use snd_soc_update_bits for read-modify-write ASoC: wm8400: Fix wrong bit setting for WM8400_POWER_MANAGEMENT_2 ASoC: wm8990: Fix wrong bit setting for WM8990_POWER_MANAGEMENT_2 ASoC: wm8990: Use snd_soc_update_bits for read-modify-write ASoC: ad193x: Fix define of AD193X_PLL_INPUT_MASK ASoC: wm8741: Fix setting interface format for DSP modes ASoC: wm8741: Use snd_soc_cache_sync to sync reg_cache with the hardware ASoC: wm8900: Fix wrong mask for setting DAC_CLKDIV/ADC_CLKDIV/LRCLK_MODE ASoC: wm8900: Fix the mask defines ASoC: wm8900: Use snd_soc_update_bits for read-modify-write ASoC: wm8580: Use snd_soc_update_bits for read-modify-write ASoC: ad193x: Use snd_soc_update_bits for read-modify-write ASoC: sgtl5000: Fix define for SGTL5000_BIAS_R_MASK ASoC: sgtl5000: Fix setting mic bias resistor ASoC: ssm2602: Use snd_soc_update_bits for read-modify-write ASoC: ak4641: Remove unused codec field from struct ak4641_priv ASoC: ak4641: Use SND_SOC_DAPM_DAC for Voice Playback stream widget Kuninori Morimoto (1): ASoC: ak4642: fixup cache register table Lars-Peter Clausen (1): MAINTAINERS: Add maintainer for Analog Devices sound CODECs Mark Brown (7): ASoC: Update WM5100 accessory detection for revision A ASoC: Convert DA7210 to table based DAPM init MANITAINERS: Add Cragganmore reference platform to Wolfson support ASoC: Update WM1811 DCS codes for latest evaluation results ASoC: Convert WM9081 to table based control init ASoC: Fix prefixing of DAPM controls ASoC: WM9081 interrupt status register is volatile Olof Johansson (1): ASoC: Tegra: sparse cleanup Peter Ujfalusi (6): ASoC: twl6040: Remove Capture restriction for 17.64MHz sysclk ASoC: twl6040: Remove PLL usage restrictions ASoC: twl6040: Change event ordering for Earphone driver ASoC: twl6040: Rename the Earphone Driver event handler ASoC: core: Add flag to ignore pmdown_time at pcm_close ASoC: twl6040: Request core to inline the DAPM sequence MAINTAINERS | 2 + include/sound/soc.h | 1 + sound/soc/codecs/ad193x.c | 31 ++--- sound/soc/codecs/ad193x.h | 2 +- sound/soc/codecs/ak4535.c | 10 +- sound/soc/codecs/ak4641.c | 3 +- sound/soc/codecs/ak4642.c | 22 ++-- sound/soc/codecs/alc5623.c | 4 +- sound/soc/codecs/da7210.c | 264 +++++++++++++++++++++++++++++++++----- sound/soc/codecs/sgtl5000.c | 9 +- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/ssm2602.c | 30 +++-- sound/soc/codecs/sta32x.c | 33 ++--- sound/soc/codecs/tlv320aic23.c | 167 ++++++++----------------- sound/soc/codecs/tlv320aic32x4.c | 61 ++++------ sound/soc/codecs/tlv320aic3x.c | 5 +- sound/soc/codecs/twl6040.c | 54 ++------ sound/soc/codecs/wm5100.c | 3 + sound/soc/codecs/wm5100.h | 7 + sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8580.c | 22 ++-- sound/soc/codecs/wm8741.c | 14 +-- sound/soc/codecs/wm8900.c | 115 +++++++---------- sound/soc/codecs/wm8961.c | 4 +- sound/soc/codecs/wm8990.c | 69 ++++------ sound/soc/codecs/wm8991.c | 22 ++-- sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm9081.c | 5 +- sound/soc/pxa/poodle.c | 1 - sound/soc/pxa/spitz.c | 1 - sound/soc/soc-core.c | 6 +- sound/soc/soc-pcm.c | 15 ++- sound/soc/tegra/tegra_das.c | 4 +- sound/soc/tegra/tegra_pcm.c | 2 +- 34 files changed, 529 insertions(+), 467 deletions(-)

3 4

[alsa-devel] [PATCH v5] ASoC:Add support for cs42l73 codec
by Brian Austin 20 Oct '11

20 Oct '11

Signed-off-by:Brian Austin <brian.austin(a)cirrus.com> Signed-off-by:georgi Vlaev <joe(a)nucleusys.com> This patch adds support for Cirrus Logic CS42L73 low power stereo codec --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1070 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 ++++++++++ 4 files changed, 1303 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..4f0ce61 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -175,6 +176,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..b271eb5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..6dcc947 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1070 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com> + * Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.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 "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + u32 sysclk; + u8 mclksel; + u32 mclk; + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[] = { +/*0*/ 0x00, 0x42, 0xA7, 0x30, +/*4*/ 0x00, 0x00, 0xF1, 0xDF, +/*8*/ 0x3F, 0x57, 0x53, 0x00, +/*C*/ 0x00, 0x15, 0x00, 0x15, +/*A*/ 0x00, 0x15, 0x00, 0x06, +/*E*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +/*60*/ 0x00, 0x00 +}; + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mixer_text[] = { + "Left", "Right", "Mono Mix"}; + +static const struct soc_enum spk_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum spk_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new hp_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new lo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1); + +static const struct snd_kcontrol_new spk_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new spklo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new ear_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, + 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, + 6, 1, 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 1, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum), + SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum), + + SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum), + SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum), + + SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum), + SOC_ENUM("XSP Output Mixer Select", xspout_mixer_enum), + +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), + + SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1), + + SND_SOC_DAPM_PGA("DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl), + SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, + &lo_amp_ctl), + SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl), + SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl), + SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUT"), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + + {"HPOUT", NULL, "HP Amp"}, + {"LINEOUT", NULL, "LO Amp"}, + {"SPKOUT", NULL, "SPK Amp"}, + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"HP Amp", "Switch", "DAC1"}, + {"LO Amp", "Switch", "DAC1"}, + {"SPK Amp", "Switch", "DAC2"}, + {"SPKLO Amp", "Switch", "DAC2"}, + {"EAR Amp", "Switch", "DAC2"}, + + {"DAC1", NULL, "ASPIN"}, + {"DAC1", NULL, "XSPIN"}, + {"DAC1", NULL, "VSPIN"}, + + {"DAC2", NULL, "ASPIN"}, + {"DAC2", NULL, "XSPIN"}, + {"DAC2", NULL, "VSPIN"}, + + {"PGA Mux Left", "Line A", "LINEINA"}, + {"PGA Mux Right", "Line B", "LINEINB"}, + + {"PGA Mux Left", "Mic 1", "MIC1"}, + {"PGA Mux Right", "Mic 2", "MIC2"}, + + {"PGA Left", NULL, "PGA Mux Left"}, + {"PGA Right", NULL, "PGA Mux Right"}, + + {"ADC Left", "ADC", "PGA Left"}, + {"ADC Right", "ADC", "PGA Right"}, + + {"XSPOUT", NULL, "ADC Left"}, + {"XSPOUT", NULL, "ADC Right"}, + + {"ASPOUT", NULL, "ADC Left"}, + {"ASPOUT", NULL, "ADC Right"}, + + {"VSPOUT", NULL, "ADC Left"}, + {"VSPOUT", NULL, "ADC Right"}, + +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format in slave mode only\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + + } else { + /* CS42L73 Slave */ + dev_dbg(codec->dev, "DAI[%d]: Slave\n", id); + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + cs42l73_update_asrc(codec, id, srate); + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), + 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = snd_soc_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = snd_soc_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = snd_soc_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = snd_soc_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = ARRAY_SIZE(cs42l73_reg), + .reg_cache_default = cs42l73_reg, + .reg_word_size = sizeof(u8), + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..188f06d --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Brian Austin <brian.austin(a)cirrus.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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- 1.7.6.3

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[alsa-devel] tegra_wm8903 dapm_nc_pins
by Julian Scheel 20 Oct '11

20 Oct '11

Hi, is there any special reason for the dapm_nc_pins being hardcoded into tegra_wm8903.c instead of being set through the platform data? We are currently making up a platform setting for our devices and would like to have all relevant settings within the platform driver instead of having to hack the tegra_wm8903-driver with further machine_is_XX() checks. So I would suggest to add an entry to the platform data which contains a list of NC-pins which are then set to nc by the tegra_wm8903 driver on startup. Regards, Julian

3 4

[alsa-devel] Alsa Architecture doc
by Srinivas 20 Oct '11

20 Oct '11

Hi , I am new to this list also to ALSA . I have done programming using alsa user space library, But want to understand the architecture of alsa , Is there any doc available which explains what is plugin, driver, Interface ? Thanks in advance -- Regards, Srinivas

1 0

Re: [alsa-devel] [PATCH 1/1] intel8x0: Improve performance in virtual environment
by Takashi Iwai 20 Oct '11

20 Oct '11

At Thu, 20 Oct 2011 19:02:50 +0400, Konstantin Ozerkov wrote: > > This patch intended to improve performance in virtualized environments > like Parallels Desktop or VirtualBox/QEMU (virtual ICH/AC97 audio). > > I/O access is very time-expensive operation in virtual world: VCPU > can be rescheduled and in the worst case we got more than 10ms delay on > each I/O access. > > In the original code normal loop exit rule > (old_civ == current_civ && old_picb == current_picb) was never satisfied, > because old_picb was never the same as current_picb due delay inspired by > reading current_civ. As a result loop ended by timeout and we got 10x more > I/O operations. You mean in the virtual environment, right? On the real machines, this condition can be satisfied normally because the read is much faster than the sample rate. This check is needed to avoid the broken behavior around the boundary. > To prevent glitch on buffer swapping the rule (old_civ == current_civ) is > enough. Not for real (thus less logical) machines, unfortunately. Both checks were needed in the real tests. I don't mind to implement some bypass if the driver knows that it's running in a VM. But, removing blindly would cause definitely a regression. thanks, Takashi > Experimental data from Prallels Desktop 7, RHEL6 guest (I/O ops per second): > > Original code: > In Port Counter Callback > f014 41550 fffff00000179d00 ac97_bm_read_civ+0x000 > f018 41387 fffff0000017a580 ac97_bm_read_picb+0x000 > > With patch: > In Port Counter Callback > f014 4090 fffff00000179d00 ac97_bm_read_civ+0x000 > f018 1964 fffff0000017a580 ac97_bm_read_picb+0x000 > > Signed-off-by: Konstantin Ozerkov <kozerkov(a)parallels.com> > Signed-off-by: Denis V. Lunev <den(a)parallels.com> > > diff --git a/pci/intel8x0.c b/pci/intel8x0.c > index 6a5b387..bad4a65 100644 > --- a/pci/intel8x0.c > +++ b/pci/intel8x0.c > @@ -1065,8 +1065,7 @@ static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *subs > udelay(10); > continue; > } > - if (civ == igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV) && > - ptr1 == igetword(chip, ichdev->reg_offset + ichdev->roff_picb)) > + if (civ == igetbyte(chip, ichdev->reg_offset + ICH_REG_OFF_CIV)) > break; > } while (timeout--); > ptr = ichdev->last_pos; > -- > 1.7.7 >

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Re: [alsa-devel] [PATCH 1/1] intel8x0: Improve performance in virtual environment
by Jaroslav Kysela 20 Oct '11

20 Oct '11

Date 20.10.2011 17:02, Konstantin Ozerkov wrote: > This patch intended to improve performance in virtualized environments > like Parallels Desktop or VirtualBox/QEMU (virtual ICH/AC97 audio). > > I/O access is very time-expensive operation in virtual world: VCPU > can be rescheduled and in the worst case we got more than 10ms delay on > each I/O access. > > In the original code normal loop exit rule > (old_civ == current_civ && old_picb == current_picb) was never satisfied, > because old_picb was never the same as current_picb due delay inspired by > reading current_civ. As a result loop ended by timeout and we got 10x more > I/O operations. > > To prevent glitch on buffer swapping the rule (old_civ == current_civ) is > enough. I don't think so. This code was added because we got some strange situations (values) with the real intel8x0 hardware. A better way might be to identify the virtualisation (maybe using special PCI IDs) and skip these extra checks in the driver for these devices. NAK from me. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] [PATCH] ASoC: rt5631: Remove unused codec field from struct rt5631_priv
by Axel Lin 20 Oct '11

20 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/rt5631.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 86e69f4..27a078c 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -31,7 +31,6 @@ #include "rt5631.h" struct rt5631_priv { - struct snd_soc_codec *codec; int codec_version; int master; int sysclk; @@ -1632,7 +1631,6 @@ static int rt5631_probe(struct snd_soc_codec *codec) } codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - rt5631->codec = codec; return 0; } -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: ak4671: Use snd_soc_update_bits for read-modify-write
by Axel Lin 20 Oct '11

20 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4671.c | 18 ++++++------------ 1 files changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 41e3d55..de9ff66 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -168,18 +168,15 @@ static int ak4671_out2_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u8 reg; switch (event) { case SND_SOC_DAPM_POST_PMU: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg |= AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, AK4671_MUTEN); break; case SND_SOC_DAPM_PRE_PMD: - reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); - reg &= ~AK4671_MUTEN; - snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT, + AK4671_MUTEN, 0); break; } @@ -575,15 +572,12 @@ static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int ak4671_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); - snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, - reg | AK4671_PMVCM); + snd_soc_update_bits(codec, AK4671_AD_DA_POWER_MANAGEMENT, + AK4671_PMVCM, AK4671_PMVCM); break; case SND_SOC_BIAS_OFF: snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: Fix reg_word_size for ak4104
by Axel Lin 20 Oct '11

20 Oct '11

According to the register map in datasheet, the registers are 8 bit. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4104.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index cbf0b6d..d3b29dc 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -247,7 +247,7 @@ static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, .reg_cache_size = AK4104_NUM_REGS, - .reg_word_size = sizeof(u16), + .reg_word_size = sizeof(u8), }; static int ak4104_spi_probe(struct spi_device *spi) -- 1.7.5.4

3 2

[alsa-devel] [PATCH v3 2/8] ASoC: da7210: Add support for DAPM
by Ashish Chavan 20 Oct '11

20 Oct '11

This patch adds support for DAPM covering all inputs and outputs as well as ADC and DAC. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - used supply widget for mic bias - removed mic bias hookup - assigned dapm widgets and routes in card struct - removed explicit setting of default mixer settings - removed explicit setting of ios to standby mode --- sound/soc/codecs/da7210.c | 210 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 186 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0809f32..b6e5085 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -28,6 +28,8 @@ /* DA7210 register space */ #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 +#define DA7210_STARTUP2 0x04 +#define DA7210_STARTUP3 0x05 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 #define DA7210_INMIX_L 0x0D @@ -40,6 +42,7 @@ #define DA7210_DAC_SEL 0x17 #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -165,6 +168,185 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3F, 0, hp_out_tlv), }; +/* + * DAPM Controls + */ +/* In Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), +}; + +/* In Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), +}; + +/* Out Mixer Left */ +static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), +}; + +/* Out Mixer Right */ +static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), + + SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + + /* Input Mixers */ + SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixl_controls[0], + ARRAY_SIZE(da7210_dapm_inmixl_controls)), + + SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_inmixr_controls[0], + ARRAY_SIZE(da7210_dapm_inmixr_controls)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1), + SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixl_controls[0], + ARRAY_SIZE(da7210_dapm_outmixl_controls)), + + SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0, + &da7210_dapm_outmixr_controls[0], + ARRAY_SIZE(da7210_dapm_outmixr_controls)), + + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, + &da7210_dapm_monomix_controls[0], + ARRAY_SIZE(da7210_dapm_monomix_controls)), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("OUT2"), +}; + +/* DAPM audio route definition */ +static const struct snd_soc_dapm_route da7210_audio_map[] = { + /* Dest Connecting Widget source */ + /* Input path */ + {"Mic Left", NULL, "MICL"}, + {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, + + {"In Mixer Left", "Mic Left Switch", "Mic Left"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Mic Right Switch", "Mic Right"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + + {"INPGA Left", NULL, "In Mixer Left"}, + {"ADC Left", NULL, "INPGA Left"}, + + {"INPGA Right", NULL, "In Mixer Right"}, + {"ADC Right", NULL, "INPGA Right"}, + + /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, + {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, + {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, + + {"OUTPGA Left Enable", NULL, "Out Mixer Left"}, + {"OUTPGA Right Enable", NULL, "Out Mixer Right"}, + + {"Out1 Left", NULL, "OUTPGA Left Enable"}, + {"OUT1L", NULL, "Out1 Left"}, + + {"Out1 Right", NULL, "OUTPGA Right Enable"}, + {"OUT1R", NULL, "Out1 Right"}, + + {"Headphone Left", NULL, "OUTPGA Left Enable"}, + {"HPL", NULL, "Headphone Left"}, + + {"Headphone Right", NULL, "OUTPGA Right Enable"}, + {"HPR", NULL, "Headphone Right"}, + + {"Out2 Mono", NULL, "Mono Mixer"}, + {"OUT2", NULL, "Out2 Mono"}, +}; + /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; @@ -204,29 +386,6 @@ static int da7210_volatile_register(struct snd_soc_codec *codec, return 0; } } -static int da7210_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - struct snd_soc_codec *codec = dai->codec; - - if (is_play) { - /* Enable Out */ - snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10); - snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); - - } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); - } - - return 0; -} /* * Set PCM DAI word length. @@ -409,7 +568,6 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) /* DAI operations */ static struct snd_soc_dai_ops da7210_dai_ops = { - .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, }; @@ -544,6 +702,10 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .reg_word_size = sizeof(u8), .reg_cache_default = da7210_reg, .volatile_register = da7210_volatile_register, + .dapm_widgets = da7210_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7210_dapm_widgets), + .dapm_routes = da7210_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- 1.7.1

2 8

[alsa-devel] ASoC driver parts probing order (MPC5200/MPC5121)
by David Jander 20 Oct '11

20 Oct '11

Hi all, I am writing a AC97 ASoC driver for the MPC5121e SoC from Freescale. This SoC has almost the same PSC (Programmable Serial Controllers) as the MPC5200B, for which there already is an AC97 driver: sound/soc/fsl/mpc5200-ac97.c, so I'd like to extend that one to also support the MPC5121e. This driver is supposed to work together with a second part, implementing the actual PCM driver: sound/soc/fsl/mpc5200_dma.c. Both drivers register themselves via platform_driver_register() in a regular module init function. The problem is, they do share the same driver data struct! So psc_ac97_of_probe() starts off calling psc_dma = dev_get_drvdata(&op->dev); right ahead and accessing it without checking. The DMA part of the driver does indeed create this struct psc_dma and calls dev_set_drvdata() on it at the end of the probe function. So obviously, it is supposed that the DMA driver somehow gets probed before the PSC driver, but I can't see where this is enforced. AFAIK, the order is fairly random, so it could be the other way around.... and indeed, in my case it is. Unfortunately I don't have MPC5200 hardware with AC97 to test this out, but only a MPC5121e based board. If I mimic this on the MPC5121e, it won't work, because the PSC driver gets probed first and obviously crashes with a NULL-pointer dereference in the next line of code! If this is supposed to be broken, I'd like to fix it of course, but: 1.- I can't test it on a MPC5200B, so therefor I need help. 2.- Simply turning the dev_get/set_drvdata and chip initialization order around does work in my case, but doesn't seem a robust solution to me. Any idea how this should be handled? Best regards, -- David Jander Protonic Holland.

3 6

Re: [alsa-devel] max channels of 3 jacks 6 channels motherboard using model=auto realtek codec
by Takashi Iwai 20 Oct '11

20 Oct '11

At Sun, 10 Jul 2011 08:47:46 +0200, Takashi Iwai wrote: > > At Sun, 10 Jul 2011 11:31:44 +0800, > Raymond Yau wrote: > > > > 2011/7/6 Takashi Iwai <tiwai(a)suse.de>: > > > At Wed, 6 Jul 2011 11:53:47 +0800, > > > Raymond Yau wrote: > > >> > > >> Refer to commit ce764ab22e40a046065c9417ee5f04ad2a816ac1 > > >> > > >> ALSA: hda - Add channel-mode support to Realtek auto-parser > > >> > > >> It seem that max channel is six instead of two when the channel mode > > >> is 2ch for those 3 jacks motherboard > > >> > > >> using hda-emu and alsa-info from > > >> http://thread.gmane.org/gmane.linux.alsa.devel/86816 > > >> > > >> > PCM 0 p 44100 2 16 > > >> Open PCM ALC887 Analog for play > > >> Available PCM parameters: > > >> channels: 2/6 > > >> formats: S16_LE S32_LE > > >> rates: 44100 48000 96000 192000 > > >> Prepare PCM, rate=44100, channels=2, format=16 bits > > >> > > >> > > >> because > > >> > > >> spec->multiout.max_channels = spec->multiout.num_dacs * 2; > > >> > > >> is assigned after alc_auto_add_multi_channel_mode() increase > > >> spec->multiout.num_dacs from 1 to 3 > > >> > > >> spec->multiout.num_dacs = num_pins + 1; > > > > > > Right, and this is intentional. Since there should be enough amount > > > of DACs, the driver keeps using the surround DACs statically. > > > In that way, apps can keep opening the device in the surround mode > > > while user can switch the outputs on the fly via channel-mode mixer > > > enum. > > > > > > > > > > How can the driver support 6 channels when the input jacks have not > > been retasked as output ? > > Even if the pins aren't retasked, DACs can receive data. And then > you can change the pins and routing on the fly. > > You can open a two-channel PCM stream even if the master or (any > other) volume is muted. A similar reason. > > > This behaviour seem different from model=3stack-6ch or > > model=3stack-6ch-dig or those via codec with smart51 "OFF" The smart51 case is a bit different (we prefer DAC unused because of independent HP control), but the 3stack-6ch case is a regression, indeed. This was just because of dropped code snippet. Now I fixed with the patch below. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs During the rewrite, the check of spec->need_dac_fix and the corresponding num_dacs change was dropped from the channel-mode control. This patch re-adds it, and also enables need_dac_fix for ALC880 as default, as this feature was originally introduced to fix h/w bugs of this chip. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 124c63f..52ce075 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3196,6 +3196,8 @@ static int alc_auto_ch_mode_put(struct snd_kcontrol *kcontrol, for (i = 0; i < spec->multi_ios; i++) alc_set_multi_io(codec, i, i < ch); spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix && !spec->const_channel_count) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; return 1; } @@ -3642,6 +3644,7 @@ static int patch_alc880(struct hda_codec *codec) codec->spec = spec; spec->mixer_nid = 0x0b; + spec->need_dac_fix = 1; board_config = alc_board_config(codec, ALC880_MODEL_LAST, alc880_models, alc880_cfg_tbl); -- 1.7.6

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm9081: Fix setting soft VMID ramp enable with VMID 2*240k
by Axel Lin 20 Oct '11

20 Oct '11

According to the datasheet: BIT 2:1 VMID_SEL[1:0] VMID Divider Enable and Select 00 = VMID disabled 01 = 2x40k Omh divider 10 = 2x240k Omh divider 11 = 2x5k Omh divider To set VMID 2*240k, we should OR reg with 0x04 instead of 0x40. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9081.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b2d3448..66092ef 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -819,7 +819,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, /* VMID 2*240k */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x40; + reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Standby bias current on */ -- 1.7.4.1

2 6

Re: [alsa-devel] [PATCH] ALSA: firewire: Neaten cmp_error function
by Clemens Ladisch 20 Oct '11

20 Oct '11

Joe Perches wrote: > Use an actual struct va_format instead of casting > an array of pointers. > - dev_err(&c->resources.unit->device, "%cPCR%u: %pV", > - 'i', c->pcr_index, &(struct va_format){ fmt, &va }); There is no cast. This compound literal _is_ an actual struct va_format. > Move a constant char argument to the format string. Good catch, but this module is going to be extended to handle oPCRs, too. Regards, Clemens

1 0

[alsa-devel] Writing a DMA-less PCM audio driver
by David Jander 20 Oct '11

20 Oct '11

Hi all, I am writing a PCM audio driver for a piece of (embedded) hardware that has no DMA, only a FIFO capable of holding 2048 samples. I was trying to use a kthread with high realtime (SCHED_FIFO) priority to keep the FIFO filled, and sleep with schedule_timeout(x), where "x" is depending on the current FIFO fill level. The same thread also calls snd_pcm_period_elapsed() on every completed period. It seems to work, but I get sporadic audio skips and I am trying to figure out where they come from. Most of the time schedule_timeout(1) takes almost exactly 1ms (1 jiffy at HZ=1000) to complete, but sometimes it takes up to 30ms. It seems related to file-IO (via NFS) happening on the system, but the effect is far bigger if the program that is playing the audio (mplayer) itself is producing file-IO. When using a large cache parameter on mplayer (effectively preloading the entire MP3 file into RAM), the problem is almost gone. Before deciding whether I should debug the network driver to see if it could produce such tremendous amounts of latency, is there a better way to solve this problem? Are there any other DMA-less audio drivers I could look at as an example? Any suggestion is welcome. Best regards, -- David Jander Protonic Holland.

2 3

[alsa-devel] [patch] [ALSA] hwdep: silence integer overflow warning
by Dan Carpenter 20 Oct '11

20 Oct '11

Smatch complains that if device is INT_MAX then device + 1 can overflow. It just means we would have an annoying loop while we check all the devices from -2147483648 to SNDRV_MINOR_HWDEPS. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index c7ceb28..edfdf6d 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -273,7 +273,14 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; mutex_lock(&register_mutex); - device = device < 0 ? 0 : device + 1; + + if (device < 0) + device = 0; + else if (device > SNDRV_MINOR_HWDEPS) + device = SNDRV_MINOR_HWDEPS; + else + device++; + while (device < SNDRV_MINOR_HWDEPS) { if (snd_hwdep_search(card, device)) break;

1 0

[alsa-devel] [PATCH 3/3] sound/soc/au1x: keep pointer to resource so it can be freed
by Julia Lawall 19 Oct '11

19 Oct '11

From: Julia Lawall <julia(a)diku.dk> Add a new variable for storing resources accessed subsequent to the one accessed using request_mem_region, so the one accessed using request_mem_region can be released if needed. This code is also missing some calls to iounmap. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ expression E, E1; identifier f; statement S1,S2,S3; @@ if (E == NULL) { ... when != if (E == NULL || ...) S1 else S2 when != E = E1 *E->f ... when any return ...; } else S3 // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/au1x/ac97c.c | 22 ++++++++++++---------- sound/soc/au1x/i2sc.c | 22 ++++++++++++---------- 2 files changed, 24 insertions(+), 20 deletions(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 13802ff..0188643 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -226,7 +226,7 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = { static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *r, *r1; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); @@ -249,15 +249,15 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; /* switch it on */ WR(ctx, AC97_ENABLE, EN_D | EN_CE); @@ -270,11 +270,13 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver); if (ret) - goto out1; + goto out2; ac97c_workdata = ctx; return 0; +out2: + iounmap(ctx->mmio); out1: release_mem_region(r->start, resource_size(r)); out0: diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 19e0d2a..3725c77 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -228,7 +228,7 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = { static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *r, *r1; struct au1xpsc_audio_data *ctx; ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); @@ -249,24 +249,26 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) if (!ctx->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) - goto out1; - ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) + goto out2; + ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; platform_set_drvdata(pdev, ctx); ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver); if (ret) - goto out1; + goto out2; return 0; +out2: + iounmap(ctx->mmio); out1: release_mem_region(r->start, resource_size(r)); out0:

4 6

[alsa-devel] [PATCH] ASoC: ak4641: Use SND_SOC_DAPM_DAC for Voice Playback stream widget
by Axel Lin 19 Oct '11

19 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4641.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index f5125ae..7783858 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -225,7 +225,7 @@ static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = { SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0), SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0), - SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), + SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0), SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0), SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0), -- 1.7.5.4

3 2

[alsa-devel] [PATCH] SoC: ak4641: Remove unused codec field from struct ak4641_priv
by Axel Lin 19 Oct '11

19 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4641.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7a64e58..f5125ae 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -31,7 +31,6 @@ /* codec private data */ struct ak4641_priv { - struct snd_soc_codec *codec; unsigned int sysclk; int deemph; int playback_fs; -- 1.7.5.4

3 2

[alsa-devel] [PATCH v6 2/2] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 19 Oct '11

19 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v5: - Changed error code from -EPERM to EINVAL - Used bit fields instead of magic numbers Changes since v1: - Reading from register instead of cache - Made debug message more specific about error - Changed "Gain" to "Volume" in names of ALC controls --- sound/soc/codecs/da7210.c | 110 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 110 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a31be70..f0a1999 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,11 +26,15 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A #define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -59,6 +63,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -81,6 +91,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -150,6 +161,29 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -202,6 +236,69 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EINVAL; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC | + DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC)) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EINVAL; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -260,6 +357,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

3 2

[alsa-devel] [PATCH v6 1/2] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 19 Oct '11

19 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v4: - Converted ADC mute controls to a single stereo control Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 480fc6d..a31be70 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,7 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX2 0x0B #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -41,6 +42,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -49,6 +51,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -144,6 +147,9 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -189,6 +195,13 @@ static const struct soc_enum da7210_dac_vf_cutoff = static const struct soc_enum da7210_adc_vf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -232,6 +245,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -448,6 +476,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -456,6 +496,7 @@ static struct snd_soc_dai_ops da7210_dai_ops = { .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -545,6 +586,9 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

3 2

[alsa-devel] [PATCH RESEND] ASoC: ak4641: Remove unused codec field from struct ak4641_priv
by Axel Lin 19 Oct '11

19 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Sorry. a typo in subject line. here is a resend. sound/soc/codecs/ak4641.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 7a64e58..f5125ae 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -31,7 +31,6 @@ /* codec private data */ struct ak4641_priv { - struct snd_soc_codec *codec; unsigned int sysclk; int deemph; int playback_fs; -- 1.7.5.4

1 0

[alsa-devel] ASoC: Correct way to set private data in machine driver?
by Abdullah, Omair M 19 Oct '11

19 Oct '11

Hi all, Looking at the code of soc and any machine driver, soc-core uses dev_set_drvdata() and dev_get_drvdata() to pass a snd_soc_card structure. This needs to be set in the probe to the machine driver's snd_soc_card structure. Then how can the machine driver's private data be passed around (eg. in probe and remove)? regards, -- Omair

2 1

[PATCH] ASoC: au1x: fix error paths in ac97c/i2cs probe callbacks.
by Manuel Lauss 19 Oct '11

19 Oct '11

add iounmap() and release the correct mem resource. Signed-off-by: Manuel Lauss <manuel.lauss(a)googlemail.com> --- sound/soc/au1x/ac97c.c | 7 +++++-- sound/soc/au1x/i2sc.c | 7 +++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 13802ff..4ec39aa 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -251,12 +251,12 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; /* switch it on */ @@ -275,7 +275,10 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev) ac97c_workdata = ctx; return 0; +out2: + iounmap(ctx->mmio); out1: + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(r->start, resource_size(r)); out0: kfree(ctx); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 19e0d2a..59fb31a 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -251,12 +251,12 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) r = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; r = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!r) - goto out1; + goto out2; ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; platform_set_drvdata(pdev, ctx); @@ -267,7 +267,10 @@ static int __devinit au1xi2s_drvprobe(struct platform_device *pdev) return 0; +out2: + iounmap(ctx->mmio); out1: + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(r->start, resource_size(r)); out0: kfree(ctx); -- 1.7.7

1 0

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Fix define for SGTL5000_BIAS_R_MASK
by Axel Lin 19 Oct '11

19 Oct '11

According to the datasheet: CHIP_MIC_CTRL 0x002A BITS[9:8] BIAS_RESISTOR 0x0 = Powerd off 0x1 = 2.0 kohm 0x2 = 4.0 kohm 0x3 = 8.0 kohm Thus SGTL5000_BIAS_R_MASK should be defined as 0x0300 instead of 0x0200. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sgtl5000.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index eec3ab3..8a9f435 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -280,7 +280,7 @@ /* * SGTL5000_CHIP_MIC_CTRL */ -#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_MASK 0x0300 #define SGTL5000_BIAS_R_SHIFT 8 #define SGTL5000_BIAS_R_WIDTH 2 #define SGTL5000_BIAS_R_off 0x0 -- 1.7.5.4

4 4

[alsa-devel] [PATCH] ASoC: ssm2602: Use snd_soc_update_bits for read-modify-write
by Axel Lin 19 Oct '11

19 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ssm2602.c | 30 ++++++++++++++++-------------- 1 files changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index e149ec6..3cb3271 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -343,12 +343,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, static int ssm2602_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE; + if (mute) - snd_soc_write(codec, SSM2602_APDIGI, - mute_reg | APDIGI_ENABLE_DAC_MUTE); + snd_soc_update_bits(codec, SSM2602_APDIGI, + APDIGI_ENABLE_DAC_MUTE, + APDIGI_ENABLE_DAC_MUTE); else - snd_soc_write(codec, SSM2602_APDIGI, mute_reg); + snd_soc_update_bits(codec, SSM2602_APDIGI, + APDIGI_ENABLE_DAC_MUTE, 0); return 0; } @@ -540,12 +542,12 @@ static int ssm2602_resume(struct snd_soc_codec *codec) static int ssm2602_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret, reg; + int ret; - reg = snd_soc_read(codec, SSM2602_LOUT1V); - snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH); - reg = snd_soc_read(codec, SSM2602_ROUT1V); - snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); + snd_soc_update_bits(codec, SSM2602_LOUT1V, + LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH); + snd_soc_update_bits(codec, SSM2602_ROUT1V, + ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); ret = snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); @@ -578,7 +580,7 @@ static int ssm2604_probe(struct snd_soc_codec *codec) static int ssm260x_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - int ret, reg; + int ret; pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION); @@ -595,10 +597,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec) } /* set the update bits */ - reg = snd_soc_read(codec, SSM2602_LINVOL); - snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH); - reg = snd_soc_read(codec, SSM2602_RINVOL); - snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH); + snd_soc_update_bits(codec, SSM2602_LINVOL, + LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH); + snd_soc_update_bits(codec, SSM2602_RINVOL, + RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH); /*select Line in as default input*/ snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); -- 1.7.5.4

3 2

[alsa-devel] [PATCH v2] ASoC: ad193x: Use snd_soc_update_bits for read-modify-write
by Axel Lin 19 Oct '11

19 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ad193x.c | 31 +++++++++++++++---------------- 1 files changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 39056ce..1206021 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -103,12 +103,14 @@ static const struct snd_soc_dapm_route audio_paths[] = { static int ad193x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - int reg; - reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (mute > 0) ? reg | AD193X_DAC_MASTER_MUTE : reg & - (~AD193X_DAC_MASTER_MUTE); - snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + if (mute) + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_MASTER_MUTE, + AD193X_DAC_MASTER_MUTE); + else + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_MASTER_MUTE, 0); return 0; } @@ -262,7 +264,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int word_len = 0, reg = 0, master_rate = 0; + int word_len = 0, master_rate = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; @@ -297,18 +299,15 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } - reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); - reg = (reg & (~AD193X_PLL_INPUT_MASK)) | master_rate; - snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); + snd_soc_update_bits(codec, AD193X_PLL_CLK_CTRL0, + AD193X_PLL_INPUT_MASK, master_rate); - reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) - | (word_len << AD193X_DAC_WORD_LEN_SHFT); - snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + snd_soc_update_bits(codec, AD193X_DAC_CTRL2, + AD193X_DAC_WORD_LEN_MASK, + word_len << AD193X_DAC_WORD_LEN_SHFT); - reg = snd_soc_read(codec, AD193X_ADC_CTRL1); - reg = (reg & (~AD193X_ADC_WORD_LEN_MASK)) | word_len; - snd_soc_write(codec, AD193X_ADC_CTRL1, reg); + snd_soc_update_bits(codec, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } -- 1.7.5.4

4 3

[alsa-devel] [PATCH v5 2/2] ASoC: da7210: Add support for ALC and Noise suppression
by Ashish Chavan 19 Oct '11

19 Oct '11

This patch adds controls to set following ALC parameters, - Max gain, Min gain, Noise gain, Attack rate, Release rate and delay It also adds a switch to enable/disable noise suppression. As per DA7210 data sheet, ALC and noise suppression can be enabled only if certain conditions are met. This condition checks are handled by simply using "_EXT" version of controls to capture change events. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Reading from register instead of cache - Made debug message more specific about error - Changed "Gain" to "Volume" in names of ALC controls --- sound/soc/codecs/da7210.c | 109 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 109 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a31be70..edcfc36 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -26,11 +26,15 @@ #include <sound/tlv.h> /* DA7210 register space */ +#define DA7210_CONTROL 0x01 #define DA7210_STATUS 0x02 #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A #define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -59,6 +63,12 @@ #define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_ALC_MAX 0x83 +#define DA7210_ALC_MIN 0x84 +#define DA7210_ALC_NOIS 0x85 +#define DA7210_ALC_ATT 0x86 +#define DA7210_ALC_REL 0x87 +#define DA7210_ALC_DEL 0x88 #define DA7210_A_HID_UNLOCK 0x8A #define DA7210_A_TEST_UNLOCK 0x8B #define DA7210_A_PLL1 0x90 @@ -81,6 +91,7 @@ #define DA7210_IN_R_EN (1 << 7) /* ADC bit fields */ +#define DA7210_ADC_ALC_EN (1 << 0) #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) @@ -150,6 +161,29 @@ /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) +/* CONTROL bit fields */ +#define DA7210_NOISE_SUP_EN (1 << 3) + +/* IN_GAIN bit fields */ +#define DA7210_INPGA_L_VOL (0x0F << 0) +#define DA7210_INPGA_R_VOL (0xF0 << 0) + +/* ZERO_CROSS bit fields */ +#define DA7210_AUX1_L_ZC (1 << 0) +#define DA7210_AUX1_R_ZC (1 << 1) +#define DA7210_HP_L_ZC (1 << 6) +#define DA7210_HP_R_ZC (1 << 7) + +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_VOL (0x3F << 0) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_VOL (0x3F << 0) + +/* Minimum INPGA and AUX1 volume to enable noise suppression */ +#define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ +#define DA7210_AUX1_MIN_VOL_NS 0x35 /* 6dB */ + #define DA7210_VERSION "0.0.1" /* @@ -202,6 +236,68 @@ static const char *da7210_hp_mode_txt[] = { static const struct soc_enum da7210_hp_mode_sel = SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +/* ALC can be enabled only if noise suppression is disabled */ +static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0]) { + /* Check if noise suppression is enabled */ + if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) { + dev_dbg(codec->dev, + "Disable noise suppression to enable ALC\n"); + return -EPERM; + } + } + /* If all conditions are met or we are actually disabling ALC */ + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* Noise suppression can be enabled only if following conditions are met + * ALC disabled + * ZC enabled for HP and AUX1 PGA + * INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB + * AUX1_L_VOL and AUX1_R_VOL >= 6 dB + */ +static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u8 val; + + if (ucontrol->value.integer.value[0]) { + /* Check if ALC is enabled */ + if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN) + goto err; + + /* Check ZC for HP and AUX1 PGA */ + if ((snd_soc_read(codec, DA7210_ZERO_CROSS) & + (DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | + DA7210_HP_R_ZC)) != 0xc3) + goto err; + + /* Check INPGA_L_VOL and INPGA_R_VOL */ + val = snd_soc_read(codec, DA7210_IN_GAIN); + if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) || + (((val & DA7210_INPGA_R_VOL) >> 4) < + DA7210_INPGA_MIN_VOL_NS)) + goto err; + + /* Check AUX1_L_VOL and AUX1_R_VOL */ + if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) < + DA7210_AUX1_MIN_VOL_NS) || + ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) < + DA7210_AUX1_MIN_VOL_NS)) + goto err; + } + /* If all conditions are met or we are actually disabling Noise sup */ + return snd_soc_put_volsw(kcontrol, ucontrol); + +err: + return -EPERM; +} + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -260,6 +356,19 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), SOC_ENUM("Headphone Class", da7210_hp_mode_sel), + + /* ALC controls */ + SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0, + snd_soc_get_volsw, da7210_put_alc_sw), + SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0), + SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0), + SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0), + + SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1, + 0, snd_soc_get_volsw, da7210_put_noise_sup_sw), }; /* Codec private data */ -- 1.7.1

2 2

[alsa-devel] [PATCH v4] ASoC:Add support for cs42l73 codec
by Brian Austin 19 Oct '11

19 Oct '11

Signed-off-by:Brian Austin <brian.austin(a)cirrus.com> Signed-off-by:Georgi Vlaev <joe(a)nucleusys.com> This patch adds support for Cirrus Logic CS42L73 low power stereo codec. --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l73.c | 1070 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 227 ++++++++++ 4 files changed, 1313 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4584514..22ab2fc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L52 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -175,6 +177,16 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L52 + tristate + +config SND_SOC_CS42L73 + tristate + +config SND_SOC_CS42L73_MIC2SDET + bool + depends on SND_SOC_CS42L73 + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a2c7842..5e6522d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,8 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l52-objs := cs42l52.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -115,6 +117,8 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..6dcc947 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1070 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com> + * Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.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. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/slab.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 "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; +struct cs42l73_private { + u32 sysclk; + u8 mclksel; + u32 mclk; + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[] = { +/*0*/ 0x00, 0x42, 0xA7, 0x30, +/*4*/ 0x00, 0x00, 0xF1, 0xDF, +/*8*/ 0x3F, 0x57, 0x53, 0x00, +/*C*/ 0x00, 0x15, 0x00, 0x15, +/*A*/ 0x00, 0x15, 0x00, 0x06, +/*E*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +/*60*/ 0x00, 0x00 +}; + +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14, 75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2500, 0); + +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = + SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = + SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +static const char * const cs42l73_ng_delay_text[] = { + "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +static const char * const charge_pump_freq_text[] = { + "0", "1", "2", "3", "4", + "5", "6", "7", "8", "9", + "10", "11", "12", "13", "14", "15" }; + +static const struct soc_enum charge_pump_enum = + SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, + ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); + +static const char * const cs42l73_mono_mixer_text[] = { + "Left", "Right", "Mono Mix"}; + +static const struct soc_enum spk_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum spk_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_asp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const struct soc_enum esl_xsp_mono_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0, + ARRAY_SIZE(cs42l73_mono_mixer_text), cs42l73_mono_mixer_text); + +static const char * const cs42l73_ip_swap_text[] = { + "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; + +static const struct soc_enum vspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct soc_enum xspout_mixer_enum = + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); + +static const struct snd_kcontrol_new hp_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 0, 1, 1); + +static const struct snd_kcontrol_new lo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 1, 1, 1); + +static const struct snd_kcontrol_new spk_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 2, 1, 1); + +static const struct snd_kcontrol_new spklo_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 4, 1, 1); + +static const struct snd_kcontrol_new ear_amp_ctl = + SOC_DAPM_SINGLE("Switch", CS42L73_PWRCTL3, 3, 1, 1); + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { + SOC_DOUBLE_R_SX_TLV("Headphone Analog Playback Volume", + CS42L73_HPAAVOL, CS42L73_HPBAVOL, 7, + 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 0xffffffC1, 0x0C, hpaloa_tlv), + + SOC_DOUBLE_R_SX_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 5, 0xffffff35, + 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_SX_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 7, 0xffffffA0, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_SX_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 7, 0xffffffE5, + 0xE4, hl_tlv), + + SOC_SINGLE_TLV("ADC A Boost Volume", + CS42L73_ADCIPC, 2, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("ADC B Boost Volume", + CS42L73_ADCIPC, 6, 0x01, 1, adc_boost_tlv), + + SOC_SINGLE_TLV("Speakerphone Digital Playback Volume", + CS42L73_SPKDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_SINGLE_TLV("Ear Speaker Digital Playback Volume", + CS42L73_ESLDVOL, 0, 0xE4, 1, hl_tlv), + + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + + SOC_DOUBLE("ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + + SOC_SINGLE("HL Limiter Attack Rate", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + + + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, limiter_tlv), + + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, + 6, 1, 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, limiter_tlv), + + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 1, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1, + limiter_tlv), + + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + + SOC_DOUBLE_R_TLV("XSP-IP Volume", + CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-XSP Volume", + CS42L73_XSPAXSPAA, CS42L73_XSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-ASP Volume", + CS42L73_XSPAASPAA, CS42L73_XSPAASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("XSP-VSP Volume", + CS42L73_XSPAVSPMA, CS42L73_XSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("ASP-IP Volume", + CS42L73_ASPAIPAA, CS42L73_ASPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-XSP Volume", + CS42L73_ASPAXSPAA, CS42L73_ASPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-ASP Volume", + CS42L73_ASPAASPAA, CS42L73_ASPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("ASP-VSP Volume", + CS42L73_ASPAVSPMA, CS42L73_ASPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("VSP-IP Volume", + CS42L73_VSPAIPAA, CS42L73_VSPBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-XSP Volume", + CS42L73_VSPAXSPAA, CS42L73_VSPBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-ASP Volume", + CS42L73_VSPAASPAA, CS42L73_VSPBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("VSP-VSP Volume", + CS42L73_VSPAVSPMA, CS42L73_VSPBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_DOUBLE_R_TLV("HL-IP Volume", + CS42L73_HLAIPAA, CS42L73_HLBIPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-XSP Volume", + CS42L73_HLAXSPAA, CS42L73_HLBXSPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-ASP Volume", + CS42L73_HLAASPAA, CS42L73_HLBASPBA, 0, 0x3F, 1, + attn_tlv), + SOC_DOUBLE_R_TLV("HL-VSP Volume", + CS42L73_HLAVSPMA, CS42L73_HLBVSPMA, 0, 0x3F, 1, + attn_tlv), + + SOC_SINGLE_TLV("SPK-IP Mono Volume", + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-XSP Mono Volume", + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-ASP Mono Volume", + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("SPK-VSP Mono Volume", + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_SINGLE_TLV("ESL-IP Mono Volume", + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-XSP Mono Volume", + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-ASP Mono Volume", + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), + SOC_SINGLE_TLV("ESL-VSP Mono Volume", + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), + + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + + SOC_ENUM("ESL-XSP Mono Mixer Select", esl_xsp_mono_mixer_enum), + SOC_ENUM("ESL-ASP Mono Mixer Select", esl_asp_mono_mixer_enum), + + SOC_ENUM("SPK-ASP Mono Mixer Select", spk_asp_mono_mixer_enum), + SOC_ENUM("SPK-XSP Mono Mixer Select", spk_xsp_mono_mixer_enum), + + SOC_ENUM("VSP Output Mixer Select", vspout_mixer_enum), + SOC_ENUM("XSP Output Mixer Select", xspout_mixer_enum), + +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS42L73_PWRCTL2, 6, 1, NULL, 0), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), + + SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0, + CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0, + CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0, + CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0, + CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1), + + SND_SOC_DAPM_PGA("DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux), + + SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl), + SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, + &lo_amp_ctl), + SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl), + SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl), + SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl), + + SND_SOC_DAPM_OUTPUT("HPOUT"), + SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route cs42l73_audio_map[] = { + + {"HPOUT", NULL, "HP Amp"}, + {"LINEOUT", NULL, "LO Amp"}, + {"SPKOUT", NULL, "SPK Amp"}, + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"HP Amp", "Switch", "DAC1"}, + {"LO Amp", "Switch", "DAC1"}, + {"SPK Amp", "Switch", "DAC2"}, + {"SPKLO Amp", "Switch", "DAC2"}, + {"EAR Amp", "Switch", "DAC2"}, + + {"DAC1", NULL, "ASPIN"}, + {"DAC1", NULL, "XSPIN"}, + {"DAC1", NULL, "VSPIN"}, + + {"DAC2", NULL, "ASPIN"}, + {"DAC2", NULL, "XSPIN"}, + {"DAC2", NULL, "VSPIN"}, + + {"PGA Mux Left", "Line A", "LINEINA"}, + {"PGA Mux Right", "Line B", "LINEINB"}, + + {"PGA Mux Left", "Mic 1", "MIC1"}, + {"PGA Mux Right", "Mic 2", "MIC2"}, + + {"PGA Left", NULL, "PGA Mux Left"}, + {"PGA Right", NULL, "PGA Mux Right"}, + + {"ADC Left", "ADC", "PGA Left"}, + {"ADC Right", "ADC", "PGA Right"}, + + {"XSPOUT", NULL, "ADC Left"}, + {"XSPOUT", NULL, "ADC Right"}, + + {"ASPOUT", NULL, "ADC Left"}, + {"ASPOUT", NULL, "ADC Right"}, + + {"VSPOUT", NULL, "ADC Left"}, + {"VSPOUT", NULL, "ADC Right"}, + +}; + +struct cs42l73_mclk_div { + u32 mclk; + u32 srate; + u8 mmcc; +}; + +static struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs42l73_mclkx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +static int cs42l73_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs42l73_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = (priv->mclksel << 4) | + (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + snd_soc_write(codec, CS42L73_DMMCC, dmmcc); + + priv->sysclk = mclkx_coeff; + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case CS42L73_CLKID_MCLK1: + break; + case CS42L73_CLKID_MCLK2: + break; + default: + return -EINVAL; + } + + if ((cs42l73_set_mclk(dai, freq)) < 0) { + dev_err(codec->dev, "Unable to set MCLK for dai %s\n", + dai->name); + return -EINVAL; + } + + priv->mclksel = clk_id; + + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + u8 id = codec_dai->id; + u8 inv, format; + u8 spc, mmcc; + + spc = snd_soc_read(codec, CS42L73_SPC(id)); + mmcc = snd_soc_read(codec, CS42L73_MMCC(id)); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~SPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format in slave mode only\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= SPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & SPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + switch (format) { + case SND_SOC_DAIFMT_DSP_B: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE0 << 4); + if (inv == SND_SOC_DAIFMT_IB_NF) + spc |= (PCM_MODE1 << 4); + break; + case SND_SOC_DAIFMT_DSP_A: + if (inv == SND_SOC_DAIFMT_IB_IF) + spc |= (PCM_MODE1 << 4); + break; + default: + return -EINVAL; + } + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return i + 1; + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0x0f, spfs); + break; + case CS42L73_ASP: + snd_soc_update_bits(codec, CS42L73_ASPC, 0x3c, spfs << 2); + break; + case CS42L73_VSP: + snd_soc_update_bits(codec, CS42L73_VXSPFS, 0xf0, spfs << 4); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l73_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + + } else { + /* CS42L73 Slave */ + dev_dbg(codec->dev, "DAI[%d]: Slave\n", id); + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= MCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + cs42l73_update_asrc(codec, id, srate); + snd_soc_write(codec, CS42L73_SPC(id), priv->config[id].spc); + snd_soc_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret < 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + return snd_soc_update_bits(codec, CS42L73_SPC(id), + 0x7F, tristate << 7); +} + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_tristate = cs42l73_set_tristate, +}; + +static struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS, + }, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = snd_soc_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = snd_soc_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = snd_soc_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = snd_soc_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = ARRAY_SIZE(cs42l73_reg), + .reg_cache_default = cs42l73_reg, + .reg_word_size = sizeof(u8), + .dapm_widgets = cs42l73_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42l73_dapm_widgets), + .dapm_routes = cs42l73_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs42l73_audio_map), + .controls = cs42l73_snd_controls, + .num_controls = ARRAY_SIZE(cs42l73_snd_controls), +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc((sizeof *cs42l73), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, + ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..188f06d --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,227 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev <joe(a)nucleusys.com> + * Brian Austin <brian.austin(a)cirrus.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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. Class H Ctl. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, MIC2 SDT */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Ctl. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Ctl. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* HP/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* HP/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Spkphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Spkphone LO [B] Digital */ +#define CS42L73_HPAAVOL 0x1E /* HP A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* HP B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Lmtr Attack Rate HP/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Lmtr Ctl, Rel.Rate HP/Line. */ +#define CS42L73_LMAXHL 0x28 /* Lmtr Thresholds HP/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Lmtr Attack Rate Spkphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Lmtr Ctl,Release Rate Spk. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Lmtr Thresholds Spkphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Lmtr Attack Rate */ +#define CS42L73_LIMRRATEESL 0x2D /* Lmtr Ctl,Release Rate */ +#define CS42L73_LMAXESL 0x2E /* Lmtr Thresholds */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: L. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: R. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP L */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP R */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP L */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP R */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Left */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Right */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP L */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP R */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP L */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP R */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Left */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Right */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP L */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP R */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP L */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP R */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Left */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Right */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP L */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP R */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP */ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define SP_3ST (1 << 7) +#define SPDIF_I2S 0 +#define SPDIF_PCM (1 << 6) +#define PCM_MODE0 0 +#define PCM_MODE1 1 +#define PCM_MODE2 2 +#define PCM_BO_MSBLSB 0 +#define PCM_BO_LSBMSB 1 +#define MCK_SCLK_64FS 0 +#define MCK_SCLK_MCLK 2 +#define MCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Analog Mute */ +#define HPA_MUTE (1 << 7) +/* LO A/B Analog Mute */ +#define LOA_MUTE (1 << 7) +/* Digital Mute */ +#define HLAD_MUTE (1 << 0) +#define HLBD_MUTE (1 << 1) +#define SPKD_MUTE (1 << 2) +#define ESLD_MUTE (1 << 3) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- 1.7.6.3

3 2

[alsa-devel] [PATCH] ASoC: WM9081 interrupt status register is volatile
by Mark Brown 19 Oct '11

19 Oct '11

Not that we have interrupt handling in the driver at the minute. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 2b6a75c..8176138 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -173,6 +173,7 @@ static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int re { switch (reg) { case WM9081_SOFTWARE_RESET: + case WM9081_INTERRUPT_STATUS: return 1; default: return 0; -- 1.7.6.3

2 1

[alsa-devel] What is wrong in this code.
by Anders Gnistrup 19 Oct '11

19 Oct '11

Hello I have some code, which I just can figure out why it is not working. I have tried ti varify this using amixer. agn@agn-HP-EliteBook-2540p:~/laerdal/audio-pulse2alsa/system/vs2-system/bu/var/lib/alsa$ amixer -Dhw:0 cget numid=23 numid=23,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=100,200 | dBscale-min=-51.00dB,step=0.20dB,mute=0 Ofcourse I can use the "cset" command to set the volumes. No problem. Now, I would like to do this in code. Listed below. The code works just as expected when using the: snd_ctl_elem_id_set_name(id, "PCM Playback Volume"); Should be similar to "amixer -c0 cset name='PCM Playback Volume' 100,200" But when I use the snd_ctl_elem_id_set_numid(id, numid); (with numid=23) it fails. snd_hctl_find_elem simply does not find the element! The version I use is 1.0.24. I have looked into the amixer.c source code, and it works. So I gees i am just missing a tiny option to make it work, but I hust can't figure out which one! #include <errno.h> #include <assert.h> #include <alsa/asoundlib.h> #include <sys/poll.h> int main() { snd_hctl_t *hctl; snd_ctl_elem_id_t *id; int err; unsigned int numid = 23; err = snd_hctl_open(&hctl, "hw:0", 0); err = snd_hctl_load(hctl); snd_ctl_elem_id_alloca(&id); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); //snd_ctl_elem_id_set_numid(id, numid); snd_ctl_elem_id_set_name(id, "PCM Playback Volume"); snd_hctl_elem_t *elem = snd_hctl_find_elem(hctl, id); if (elem != NULL) { snd_ctl_elem_value_t *control; snd_ctl_elem_value_alloca(&control); snd_ctl_elem_value_set_id(control, id); snd_ctl_elem_value_set_integer(control, 0, 100); err = snd_hctl_elem_write(elem, control); snd_ctl_elem_value_set_integer(control, 1, 200); err = snd_hctl_elem_write(elem, control); } else printf("Cound not find elem.\n"); snd_hctl_close(hctl); }

2 3

Re: [alsa-devel] [REGRESSION] Sound goes too fast due to 798cb7e897210
by Takashi Iwai 18 Oct '11

18 Oct '11

At Tue, 18 Oct 2011 10:38:06 +0200, Éric Piel wrote: > > Op 18-10-11 10:23, Takashi Iwai schreef: > > At Tue, 18 Oct 2011 10:10:14 +0200, > > Éric Piel wrote: > : > > If that commit affects, the best fix would be to give a quirk specific > > to your device. Try to pass either position_fix=1 or position_fix=2. > > Only one of them should work (likely 2). > > > > After checking it, you can add it to position_fix_list[] in > > sound/pci/hda/hda_intel.c together with PCI SSID. > > Hello, > Thanks for the quick response. Indeed forcing position_fix=1 does fix > the bug. I'll not try to make a patch for my device. Hm, interesting. So, in your case, the position-buffer exists and reports some valid values, but the values are sloppy actually. It's hard to detect in the driver, unfortunately. The relevant commit (and its original fix) were the attempts to detect better, but it seems that it fails... FWIW, the patch below is what I'm committing to the tree. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Add position_fix quirk for Dell Inspiron 1010 The previous fix for the position-buffer check gives yet another regression on a Dell laptop. The safest fix right now is to add a static quirk for this device (and better to apply it for stable kernels too). Reported-by: Éric Piel <Eric.Piel(a)tremplin-utc.net> Cc: <stable(a)kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e9a2a87..191284a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2370,6 +2370,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1028, 0x02c6, "Dell Inspiron 1010", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), -- 1.7.7

2 4

[alsa-devel] [PATCH 1/3] sound/soc/mxs/mxs-saif.c: keep pointer to resource so it can be freed
by Julia Lawall 18 Oct '11

18 Oct '11

From: Julia Lawall <julia(a)diku.dk> Add a new variable for storing resources accessed subsequent to the one accessed using request_mem_region, so the one accessed using request_mem_region can be released if needed. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ expression E, E1; identifier f; statement S1,S2,S3; @@ if (E == NULL) { ... when != if (E == NULL || ...) S1 else S2 when != E = E1 *E->f ... when any return ...; } else S3 // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/mxs/mxs-saif.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 401944c..e12ff78 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -617,7 +617,7 @@ static irqreturn_t mxs_saif_irq(int irq, void *dev_id) static int mxs_saif_probe(struct platform_device *pdev) { - struct resource *res; + struct resource *res, *res1; struct mxs_saif *saif; struct mxs_saif_platform_data *pdata; int ret = 0; @@ -676,14 +676,14 @@ static int mxs_saif_probe(struct platform_device *pdev) goto failed_ioremap; } - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { + res1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res1) { ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); goto failed_ioremap; } - saif->dma_param.chan_num = res->start; + saif->dma_param.chan_num = res1->start; saif->irq = platform_get_irq(pdev, 0); if (saif->irq < 0) {

3 6

[alsa-devel] [PATCH v5 1/2] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 18 Oct '11

18 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v4: - Converted ADC mute controls to a single stereo control Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 480fc6d..a31be70 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,7 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX2 0x0B #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -41,6 +42,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -49,6 +51,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -144,6 +147,9 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -189,6 +195,13 @@ static const struct soc_enum da7210_dac_vf_cutoff = static const struct soc_enum da7210_adc_vf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -232,6 +245,21 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -448,6 +476,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -456,6 +496,7 @@ static struct snd_soc_dai_ops da7210_dai_ops = { .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -545,6 +586,9 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

1 0

[alsa-devel] [RFC PATCH] HDA: Generic input jack handling
by David Henningsson 18 Oct '11

18 Oct '11

So, this is what I had in mind for 3.2. Assuming positive feedback from Takashi I'll go ahead and make a real patch out of this, and to clean up the Realtek implementation, as well as probably add this method for more codecs. Thoughts: 1) The unsol event tags vary wildly between different vendors. How about standardising that as well? 2) If alc_init_jacks would call the new method, that might regress model-based (non auto) parsers. Is that a big deal these days? 3) Todo for realtek parser is to clean up all other calls to snd_input_jack_report so that jacks are not reported twice. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e9b039c..69390fd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5284,6 +5284,142 @@ int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, } EXPORT_SYMBOL_HDA(snd_hda_input_jack_add); +/** + * snd_hda_input_auto_jack_add - Add relevant input jacks based on + * auto pin configuration. + * @param unsol_tags lists of jack types to enable, terminate with + * list entry with jack_type set to 0. If unsol_tag is set to 0, + * unsol events will not be enabled for that jack type. + * @return 0 or error code + */ +int snd_hda_input_auto_jack_add(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + const struct hda_unsol_jack_tag* unsol_tags) +{ + for (; unsol_tags->jack_type; unsol_tags++) { + hda_nid_t nid_list_storage[AUTO_CFG_MAX_INS]; + const hda_nid_t *nid_list; + int nid_count = 0; + int i; + int input_match = AUTO_PIN_MIC; + + switch (unsol_tags->jack_type) { + case SND_JACK_LINEIN: + input_match = AUTO_PIN_LINE_IN; + /* Fall through */ + case SND_JACK_MICROPHONE: + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].type == input_match) + nid_list_storage[nid_count++] = cfg->inputs[i].pin; + } + nid_list = nid_list_storage; + break; + case SND_JACK_HEADPHONE: + if (cfg->line_out_type == AUTO_PIN_HP_OUT) { + nid_list = cfg->line_out_pins; + nid_count = cfg->line_outs; + } else { + nid_list = cfg->hp_pins; + nid_count = cfg->hp_outs; + } + break; + case SND_JACK_LINEOUT: + if (cfg->line_out_type == AUTO_PIN_LINE_OUT) { + nid_list = cfg->line_out_pins; + nid_count = cfg->line_outs; + } + break; + } + + for (i = 0; i < nid_count; i++) { + int pin = nid_list[i]; + int err; + if (!is_jack_detectable(codec, pin)) + continue; + + if (unsol_tags->unsol_tag) { + err = snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | unsol_tags->unsol_tag); + if (err) + return err; + } + + err = snd_hda_input_jack_add(codec, pin, + unsol_tags->jack_type, NULL); + if (err) + return err; + snd_hda_input_jack_report(codec, pin); + } + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_input_auto_jack_add); + +void snd_hda_input_jack_report_type(struct hda_codec *codec, int jack_type) +{ + struct hda_jack_item *jacks = codec->jacks.list; + int i; + + if (!jacks) + return; + + for (i = 0; i < codec->jacks.used; i++, jacks++) + if (jacks->type == jack_type) + snd_hda_input_jack_report(codec, jacks->nid); +} +EXPORT_SYMBOL_HDA(snd_hda_input_jack_report_type); + + void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) { struct hda_jack_item *jacks = codec->jacks.list; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 46c581c..bb59a3f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -678,11 +678,21 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); /* * Input-jack notification support */ + +struct hda_unsol_jack_tag { + int jack_type; /* SND_JACK_xxx constant */ + int unsol_tag; /* event tag */ +}; + #ifdef CONFIG_SND_HDA_INPUT_JACK int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, const char *name); void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid); +void snd_hda_input_jack_report_type(struct hda_codec *codec, int jack_type); void snd_hda_input_jack_free(struct hda_codec *codec); +int snd_hda_input_auto_jack_add(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + const struct hda_unsol_jack_tag *unsol_tags); #else /* CONFIG_SND_HDA_INPUT_JACK */ static inline int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, @@ -694,9 +704,19 @@ static inline void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) { } +static inline void snd_hda_input_jack_report_type(struct hda_codec *codec, + int jack_type) +{ +} static inline void snd_hda_input_jack_free(struct hda_codec *codec) { } +static inline int snd_hda_input_auto_jack_add(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + const struct hda_unsol_jack_tag *unsol_tags) +{ + return 0; +} #endif /* CONFIG_SND_HDA_INPUT_JACK */ #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf53663..90cdd6c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -38,6 +38,7 @@ #define ALC_DCVOL_EVENT 0x02 #define ALC_HP_EVENT 0x04 #define ALC_MIC_EVENT 0x08 +#define ALC_LINEIN_EVENT 0x10 /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -441,11 +442,21 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, * Jack-reporting via input-jack layer */ +static const struct hda_unsol_jack_tag unsol_tags[] = { + {.jack_type = SND_JACK_HEADPHONE, .unsol_tag = ALC_HP_EVENT }, + {.jack_type = SND_JACK_LINEOUT, .unsol_tag = ALC_FRONT_EVENT }, + {.jack_type = SND_JACK_MICROPHONE, .unsol_tag = ALC_MIC_EVENT }, + {.jack_type = SND_JACK_LINEIN, .unsol_tag = ALC_LINEIN_EVENT }, + {} /* Zero terminator */ +}; + /* initialization of jacks; currently checks only a few known pins */ static int alc_init_jacks(struct hda_codec *codec) { #ifdef CONFIG_SND_HDA_INPUT_JACK struct alc_spec *spec = codec->spec; + snd_hda_input_auto_jack_add(codec, &spec->autocfg, unsol_tags); +/* ; int err; unsigned int hp_nid = spec->autocfg.hp_pins[0]; unsigned int mic_nid = spec->ext_mic_pin; @@ -472,7 +483,7 @@ static int alc_init_jacks(struct hda_codec *codec) if (err < 0) return err; snd_hda_input_jack_report(codec, dock_nid); - } + }*/ #endif /* CONFIG_SND_HDA_INPUT_JACK */ return 0; } @@ -645,12 +656,18 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) switch (res) { case ALC_HP_EVENT: alc_hp_automute(codec); + snd_hda_input_jack_report_type(codec, SND_JACK_HEADPHONE); break; case ALC_FRONT_EVENT: alc_line_automute(codec); + snd_hda_input_jack_report_type(codec, SND_JACK_LINEOUT); break; case ALC_MIC_EVENT: alc_mic_automute(codec); + snd_hda_input_jack_report_type(codec, SND_JACK_MICROPHONE); + break; + case ALC_LINEIN_EVENT: + snd_hda_input_jack_report_type(codec, SND_JACK_LINEIN); break; } } -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 15

[alsa-devel] [PATCH v4 3/7] ASoC: da7210: Add support for mute and zero cross controls
by Ashish Chavan 18 Oct '11

18 Oct '11

This patch adds support for below set of controls, (1) Mute controls for MIC, AUX and ADC (2) Zero cross controls for head phone, AUX, INPGA and line out (3) Head phone mode selection - class H or G It also adds digital_mute() call back. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Added digital_mute() call back and removed control for "DAC Mute Switch" - Renamed mute control switches Changes since v1: - Changed Mic mute control to a double control - Fixed a typo --- sound/soc/codecs/da7210.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 480fc6d..cac4f26 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,7 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX2 0x0B #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -41,6 +42,7 @@ #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_SOFTMUTE 0x18 #define DA7210_DAC_EQ1_2 0x19 #define DA7210_DAC_EQ3_4 0x1A #define DA7210_DAC_EQ5 0x1B @@ -49,6 +51,7 @@ #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 +#define DA7210_ZERO_CROSS 0x24 #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 @@ -144,6 +147,9 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* SOFTMUTE bit fields */ +#define DA7210_RAMP_EN (1 << 6) + #define DA7210_VERSION "0.0.1" /* @@ -189,6 +195,13 @@ static const struct soc_enum da7210_dac_vf_cutoff = static const struct soc_enum da7210_adc_vf_cutoff = SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static const char *da7210_hp_mode_txt[] = { + "Class H", "Class G" +}; + +static const struct soc_enum da7210_hp_mode_sel = + SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -232,6 +245,22 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff), SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0), SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff), + + /* Mute controls */ + SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0), + SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0), + SOC_SINGLE("ADC Left Capture Switch", DA7210_ADC, 2, 1, 0), + SOC_SINGLE("ADC Right Capture Switch", DA7210_ADC, 6, 1, 0), + SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0), + SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0), + + /* Zero cross controls */ + SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0), + SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0), + SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0), + SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0), + + SOC_ENUM("Headphone Class", da7210_hp_mode_sel), }; /* Codec private data */ @@ -448,6 +477,18 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return 0; } +static int da7210_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB; + + if (mute) + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4); + else + snd_soc_write(codec, DA7210_DAC_HPF, mute_reg); + return 0; +} + #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -456,6 +497,7 @@ static struct snd_soc_dai_ops da7210_dai_ops = { .startup = da7210_startup, .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .digital_mute = da7210_mute, }; static struct snd_soc_dai_driver da7210_dai = { @@ -545,6 +587,9 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Enable ramp mode for DAC gain update */ + snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 8

Re: [alsa-devel] [PATCH] ALSA: hda/conexant - Add quirks for lenovo notebooks
by Takashi Iwai 18 Oct '11

18 Oct '11

At Tue, 18 Oct 2011 14:05:53 +0200, Torsten Hilbrich wrote: > > Am 18.10.2011 12:57, schrieb Takashi Iwai: > > At Tue, 18 Oct 2011 12:20:28 +0200, > > Torsten Hilbrich wrote: > >> > >> The lenovo sandy bridge notebooks (T420, T520, L520) didn't work with > >> the external sound input and output connectors on the dockig unit > >> (tested with P/N 45N5887). > >> > >> Loading snd-hda-intel with model=thinkpad solved the problem so I > >> added the accordings quirks to perform that decision automatically > >> for these specific models. > >> > >> Signed-off-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> > > > > Could you check whether model=auto with the latest kernel work? > > Basically I won't take any more the model-specific quirk but > > move all to the auto-parser as much as possible. > > I just checked with upstream v3.1-rc10, model=auto didn't solve the > problem on the Lenovo T520. OK, so it's a bug of BIOS. BIOS doesn't provide the pin for docking-station. The right fix in this case is to provide the pin config table. In sound git tree, there is a fix for Lenovo X200. A similar entry would work for you. git://github.com/tiwai/sound.git Check either topic/hda, for-next or master branch. > Here are some dmesg output with model=auto: > > snd_hda_intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 > snd_hda_intel 0000:00:1b.0: irq 45 for MSI/MSI-X > snd_hda_intel 0000:00:1b.0: setting latency timer to 64 > hda-intel: chipset global capabilities = 0x4401 > hda-intel: Clearing TCSEL > hda-intel: HDA snoop disabled, enabling ... OK > hda-intel: codec_mask = 0xb > hda-intel: codec #0 probed OK > hda-intel: codec #1 probed OK > hda-intel: codec #3 probed OK > IBM TrackPoint firmware: 0x0e, buttons: 3/3 > hda_codec: model 'auto' is selected > hda_codec: CX20590: BIOS auto-probing. > autoconfig: line_outs=1 (0x1f/0x0/0x0/0x0/0x0) type:speaker > speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > hp_outs=1 (0x19/0x0/0x0/0x0/0x0) > mono: mono_out=0x0 > inputs: Mic=0x1b Internal Mic=0x23 > > when playing some audio: > > hda-intel: azx_pcm_prepare: bufsize=0x10000, format=0x31 > hda_codec_setup_stream: NID=0x10, stream=0x5, channel=0, format=0x31 > hda_codec_setup_stream: NID=0x11, stream=0x5, channel=0, format=0x31 > hda_codec_cleanup_stream: NID=0x10 > hda_codec_cleanup_stream: NID=0x11 > > I will attach the codec#x files from /proc/asound/PCH if you need this > information. At best, please give alsa-info.sh output (run with --no-upload option). thanks, Takashi

1 0

[alsa-devel] ASoC: Amplifier behind codec?
by Wolfram Sang 18 Oct '11

18 Oct '11

Hi, I have here an amplifier (MAX9768) between codec output and the speaker, all controlled via I2C. I noticed that other amplifier drivers export a function called *_add_controls(), so the platform glue code can add the amplifier controls to the codec. Now, all drivers working like this use a single static instance of the i2c client, because there doesn't seem to be a connection between the added control elements and the additional i2c client. Is the above approach still the preferred way of doing this? Thanks, Wolfram -- Pengutronix e.K. | Wolfram Sang | Industrial Linux Solutions | http://www.pengutronix.de/ |

2 1

[PATCH] ALSA: HDA: Add new revision for ALC662
by David Henningsson 18 Oct '11

18 Oct '11

The revision 0x100300 was found for ALC662. It seems to work well with patch_alc662. Cc: stable(a)kernel.org BugLink: http://bugs.launchpad.net/bugs/877373 Tested-by: Shengyao Xue <Shengyao.xue(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f9d24c3..86ecc72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5774,6 +5774,8 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { .patch = patch_alc882 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, + { .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3", + .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, -- 1.7.5.4 --------------070301030905080206060603--

1 0

Re: [alsa-devel] [PATCH] ALSA: hda/conexant - Add quirks for lenovo notebooks
by Takashi Iwai 18 Oct '11

18 Oct '11

At Tue, 18 Oct 2011 12:20:28 +0200, Torsten Hilbrich wrote: > > The lenovo sandy bridge notebooks (T420, T520, L520) didn't work with > the external sound input and output connectors on the dockig unit > (tested with P/N 45N5887). > > Loading snd-hda-intel with model=thinkpad solved the problem so I > added the accordings quirks to perform that decision automatically > for these specific models. > > Signed-off-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> Could you check whether model=auto with the latest kernel work? Basically I won't take any more the model-specific quirk but move all to the auto-parser as much as possible. thanks, Takashi > --- > sound/pci/hda/patch_conexant.c | 3 +++ > 1 files changed, 3 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 502fc94..90a6b98 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -3110,8 +3110,11 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { > SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), > SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), > + SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT5066_THINKPAD), > + SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), > + SND_PCI_QUIRK(0x17aa, 0x21de, "Lenovo L520", CXT5066_THINKPAD), > SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), > SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), > SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), > -- > 1.7.5.4 >

1 0

Re: [alsa-devel] [REGRESSION] Sound goes too fast due to 798cb7e897210
by Takashi Iwai 18 Oct '11

18 Oct '11

At Tue, 18 Oct 2011 10:10:14 +0200, Éric Piel wrote: > > Dear Takashi, > > I have a regression here on 3.1-rc9+ causing the sound "going too fast" > with hda. Reverting 798cb7e897210 (ALSA: hda - Fix a regression of the > position-buffer check) fixes this regression for me. > > Of course, as this commit is itself a regression fix, my goat feeling is > that it's not going to be possible to directly revert it. Attached if > the output of alsa-info.sh (on the fixed kernel). Please let me know if > I can provide you any additional information or test any patch. If that commit affects, the best fix would be to give a quirk specific to your device. Try to pass either position_fix=1 or position_fix=2. Only one of them should work (likely 2). After checking it, you can add it to position_fix_list[] in sound/pci/hda/hda_intel.c together with PCI SSID. thanks, Takashi

1 0

[alsa-devel] [PATCH] ASoC: Convert WM9081 to table based control init
by Mark Brown 18 Oct '11

18 Oct '11

At least for the core controls, the optionally selected controls are still added programatically. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index f4fc03b..93f794c 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1235,8 +1235,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, WM9081_SPKPGAZC, WM9081_SPKPGAZC); - snd_soc_add_controls(codec, wm9081_snd_controls, - ARRAY_SIZE(wm9081_snd_controls)); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); @@ -1296,6 +1294,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .controls = wm9081_snd_controls, + .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), .dapm_routes = wm9081_audio_paths, -- 1.7.6.3

2 1

[alsa-devel] [PATCH] ASoC: Update WM1811 DCS codes for latest evaluation results
by Mark Brown 18 Oct '11

18 Oct '11

Evaluation of larger quantities of material has provided new DCS codes values to be applied for WM1811. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index df55fa3..044350a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3241,8 +3241,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 0: case 1: - wm8994->hubs.dcs_codes_l = -7; - wm8994->hubs.dcs_codes_r = -4; + wm8994->hubs.dcs_codes_l = -9; + wm8994->hubs.dcs_codes_r = -5; break; default: break; -- 1.7.6.3

2 1

[alsa-devel] [PATCH 2/3] sound/soc/au1x: keep pointer to resource so it can be freed
by Julia Lawall 18 Oct '11

18 Oct '11

From: Julia Lawall <julia(a)diku.dk> Add a new variable for storing resources accessed subsequent to the one accessed using request_mem_region, so the one accessed using request_mem_region can be released if needed. The semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @r@ expression E, E1; identifier f; statement S1,S2,S3; @@ if (E == NULL) { ... when != if (E == NULL || ...) S1 else S2 when != E = E1 *E->f ... when any return ...; } else S3 // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/au1x/psc-ac97.c | 14 +++++++------- sound/soc/au1x/psc-i2s.c | 14 +++++++------- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 172eefd..62e5640 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -364,7 +364,7 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) { int ret; - struct resource *r; + struct resource *r, *r1; unsigned long sel; struct au1xpsc_audio_data *wd; @@ -388,15 +388,15 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; /* configuration: max dma trigger threshold, enable ac97 */ wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 7c5ae92..621e260 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -290,7 +290,7 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = { static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) { - struct resource *r; + struct resource *r, *r1; unsigned long sel; int ret; struct au1xpsc_audio_data *wd; @@ -313,15 +313,15 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev) if (!wd->mmio) goto out1; - r = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r1->start; - r = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!r) + r1 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r1) goto out2; - wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start; + wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r1->start; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer)

1 0

[alsa-devel] [PATCH 1/2] ASoC: ad193x: Fix define of AD193X_PLL_INPUT_MASK
by Axel Lin 18 Oct '11

18 Oct '11

Current code defines AD193X_PLL_INPUT_MASK as (~0x6) which is quite different from other MASK defines. To make it consistent with other mask defines, define AD193X_PLL_INPUT_MASK as 0x6 and change the code accordingly. I think this change improves the readability. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/ad193x.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f934670..39056ce 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -298,7 +298,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, } reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0); - reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate; + reg = (reg & (~AD193X_PLL_INPUT_MASK)) | master_rate; snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 536e5f2..1507eaa 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -11,7 +11,7 @@ #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -#define AD193X_PLL_INPUT_MASK (~0x6) +#define AD193X_PLL_INPUT_MASK 0x6 #define AD193X_PLL_INPUT_256 (0 << 1) #define AD193X_PLL_INPUT_384 (1 << 1) #define AD193X_PLL_INPUT_512 (2 << 1) -- 1.7.4.1

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[alsa-devel] [PATCH v2] ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
by axel 18 Oct '11

18 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: For Initial cold start ( change bias level from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY ), the v1 does not set VMID_SEL[1:0] bits correctly for Normal bias enable & soft start off. In v2 we also clear VMID_SEL[1:0] bits for Normal bias enable & soft start off. We did clear VMID_SEL[1:0] bits in SND_SOC_BIAS_OFF, thus we know the VMID_SEL[1:0] bits status is 0b00 in this case. sound/soc/codecs/wm9081.c | 82 +++++++++++++++++++------------------------- 1 files changed, 35 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 66092ef..9c38ddb 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -766,84 +766,73 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); - + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | 0x6, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg |= WM9081_BIAS_ENA; - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_BIAS_ENA | + WM9081_VMID_SEL_MASK, + WM9081_BIAS_ENA); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, WM9081_BIAS_SRC); /* Disable VMID and biases with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK | + WM9081_BIAS_ENA, WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1242,11 +1231,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Fix prefixing of DAPM controls
by Mark Brown 18 Oct '11

18 Oct '11

We don't want to clear the prefix while we're creating the DAPM controls for the device as the prefix is applied during control creation. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 72c23c6..67724cdf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1055,13 +1055,13 @@ static int soc_post_component_init(struct snd_soc_card *card, } rtd->card = card; + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(&codec->dapm); + /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; codec->name_prefix = NULL; - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - /* do machine specific initialization */ if (!dailess && dai_link->init) ret = dai_link->init(rtd); -- 1.7.6.3

1 0

[alsa-devel] [PATCH v2] ASoC: wm8580: Use snd_soc_update_bits for read-modify-write
by Axel Lin 18 Oct '11

18 Oct '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: Fix below build warning: CC sound/soc/codecs/wm8580.o sound/soc/codecs/wm8580.c: In function 'wm8580_set_bias_level': sound/soc/codecs/wm8580.c:749: warning: unused variable 'reg' sound/soc/codecs/wm8580.c | 22 +++++++++------------- 1 files changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index b256727..8212b3c 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -430,8 +430,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Always disable the PLL - it is not safe to leave it running * while reprogramming it. */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg | pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, pwr_mask); if (!freq_in || !freq_out) return 0; @@ -449,8 +448,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8580_PLLA4 + offset, reg); /* All done, turn it on */ - reg = snd_soc_read(codec, WM8580_PWRDN2); - snd_soc_write(codec, WM8580_PWRDN2, reg & ~pwr_mask); + snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, 0); return 0; } @@ -748,7 +746,6 @@ static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) static int wm8580_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -757,20 +754,19 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Power up and get individual control of the DACs */ - reg = snd_soc_read(codec, WM8580_PWRDN1); - reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); - snd_soc_write(codec, WM8580_PWRDN1, reg); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN | + WM8580_PWRDN1_ALLDACPD, 0); /* Make VMID high impedance */ - reg = snd_soc_read(codec, WM8580_ADC_CONTROL1); - reg &= ~0x100; - snd_soc_write(codec, WM8580_ADC_CONTROL1, reg); + snd_soc_update_bits(codec, WM8580_ADC_CONTROL1, + 0x100, 0); } break; case SND_SOC_BIAS_OFF: - reg = snd_soc_read(codec, WM8580_PWRDN1); - snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); + snd_soc_update_bits(codec, WM8580_PWRDN1, + WM8580_PWRDN1_PWDN, WM8580_PWRDN1_PWDN); break; } codec->dapm.bias_level = level; -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm8900: Fix wrong mask for setting DAC_CLKDIV/ADC_CLKDIV/LRCLK_MODE
by Axel Lin 18 Oct '11

18 Oct '11

After checking the datasheet, I think what we want to do here is to clear the WM8900_REG_CLOCKING2_DAC_CLKDIV/WM8900_REG_CLOCKING2_ADC_CLKDIV/ WM8900_REG_DACCTRL_AIF_LRCLKRATE bits and then OR with div value. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I'd appreciate if someone can test this serial of patches. Thanks, Axel sound/soc/codecs/wm8900.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 082040e..b16522f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -844,17 +844,17 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai, case WM8900_DAC_CLKDIV: reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV)); + div | (reg & ~WM8900_REG_CLOCKING2_DAC_CLKDIV)); break; case WM8900_ADC_CLKDIV: reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); snd_soc_write(codec, WM8900_REG_CLOCKING2, - div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV)); + div | (reg & ~WM8900_REG_CLOCKING2_ADC_CLKDIV)); break; case WM8900_LRCLK_MODE: reg = snd_soc_read(codec, WM8900_REG_DACCTRL); snd_soc_write(codec, WM8900_REG_DACCTRL, - div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE)); + div | (reg & ~WM8900_REG_DACCTRL_AIF_LRCLKRATE)); break; default: return -EINVAL; -- 1.7.4.1

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