From 92fd918c2416404c2ec09829b25243b9a785dc9b Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Fri, 30 Mar 2012 09:52:25 +1300 Subject: ALSA: asihpi - fix return value of hpios_locked_mem_alloc() Make this function consistent with others in this module by returning 1 for error, instead of -ENOMEM (reverts function signature change from a938fb1e) Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi_internal.h | 4 ++-- sound/pci/asihpi/hpios.c | 10 +++++----- 2 files changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 8c63200..bc86cb7 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Copyright (C) 1997-2012 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned. If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle. */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, /**< memory handle */ u32 size, /**< Size in bytes to allocate */ struct pci_dev *p_os_reference diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index 87f4385..5ef4fe9 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. + Copyright (C) 1997-2012 AudioScience Inc. This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -39,11 +39,11 @@ void hpios_delay_micro_seconds(u32 num_micro_sec) } -/** Allocated an area of locked memory for bus master DMA operations. +/** Allocate an area of locked memory for bus master DMA operations. -On error, return -ENOMEM, and *pMemArea.size = 0 +If allocation fails, return 1, and *pMemArea.size = 0 */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, struct pci_dev *pdev) { /*?? any benefit in using managed dmam_alloc_coherent? */ @@ -62,7 +62,7 @@ int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, HPI_DEBUG_LOG(WARNING, "failed to allocate %d bytes locked memory\n", size); p_mem_area->size = 0; - return -ENOMEM; + return 1; } } -- cgit v1.1 From f0cdcf3ab6c62b3f774a2af15dfa01988e7a9b02 Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Fri, 30 Mar 2012 00:13:02 +0800 Subject: ASoC: sgtl5000: Enable VAG when DAC/ADC up As manual described, VAG is an internal voltage reference of DAC/ADC, So enabled it before DAC/ADC up. One more thing should care about is VAG fully ramped down requires 400ms, wait it to avoid pop. Signed-off-by: Zeng Zhaoming Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d192626..8e92fb8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, } /* - * using codec assist to small pop, hp_powerup or lineout_powerup - * should stay setting until vag_powerup is fully ramped down, - * vag fully ramped down require 400ms. + * As manual described, ADC/DAC only works when VAG powerup, + * So enabled VAG before ADC/DAC up. + * In power down case, we need wait 400ms when vag fully ramped down. */ -static int small_pop_event(struct snd_soc_dapm_widget *w, +static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { @@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), @@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, + power_vag_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), }; @@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ -- cgit v1.1 From cd1506736f3a77429f619ede817a119a7ff5f7e5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 30 Mar 2012 17:07:17 -0600 Subject: ASoC: tegra: ensure clocks are enabled when touching registers Debugfs files could be accessed any time, so explicitly enable clocks when reading registers to generate debugfs file content. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 4 ++++ sound/soc/tegra/tegra_spdif.c | 4 ++++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 33509de..2d98c92 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -79,11 +79,15 @@ static int tegra_i2s_show(struct seq_file *s, void *unused) struct tegra_i2s *i2s = s->private; int i; + clk_enable(i2s->clk_i2s); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_i2s_read(i2s, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(i2s->clk_i2s); + return 0; } diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index 475428c..9ff2c60 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -79,11 +79,15 @@ static int tegra_spdif_show(struct seq_file *s, void *unused) struct tegra_spdif *spdif = s->private; int i; + clk_enable(spdif->clk_spdif_out); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_spdif_read(spdif, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(spdif->clk_spdif_out); + return 0; } -- cgit v1.1 From e95cee0e36c671db2804f2763b547a86930061ea Mon Sep 17 00:00:00 2001 From: Martin Jansa Date: Mon, 2 Apr 2012 10:24:08 +0200 Subject: ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro * fixes sound/soc/pxa/pxa2xx-i2s.c:86:2: error: implicit declaration of function 'IOMEM' [-Werror=implicit-function-declaration] sound/soc/pxa/pxa2xx-i2s.c:86:2: error: initializer element is not constant after 23019a733bb83c8499f192fb428b7e6e81c95a34 removed IOMEM definition from arch/arm/mach-pxa/include/mach/hardware.h Signed-off-by: Martin Jansa Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 609abd5..d085837 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include -- cgit v1.1 From 1f99e44cf059d2ed43c5a0724fa738b83800f725 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 4 Apr 2012 23:28:01 -0700 Subject: ASoC: ak4642: fixup: mute needs +1 step ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute. But current settings didn't care +1 step for mute. This patch adds it Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index f8e10ce..b3e24f2 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -140,7 +140,7 @@ * min : 0xFE : -115.0 dB * mute: 0xFF */ -static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); static const struct snd_kcontrol_new ak4642_snd_controls[] = { -- cgit v1.1 From 00792ac4e0d88e82fc489a5e1c4d4435125a301c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 5 Apr 2012 09:45:51 -0300 Subject: ASoC: imx-audmux: Fix ssi port numbers in sysfs Doing a 'cat /sys/kernel/debug/audmux/ssi7' causes the following oops to be printed by the kernel: Uhandled fault: external abort on non-linefetch (0x008) at 0xf53b003c Internal error: : 8 [#1] PREEMPT Modules linked in: CPU: 0 Not tainted (3.3.0-00033-gecc726e-dirty #307) PC is at audmux_read_file+0x68/0x2f4 LR is at clk_enable+0x3c/0x48 pc : [] lr : [] psr: a0000013 sp : c3ad3f38 ip : c30a4000 fp : 00000003 r10: 00001000 r9 : be83fb00 r8 : c3ad3f80 r7 : c3ad3f80 r6 : 00000007 r5 : 00031010 r4 : c30a5000 r3 : f53b0000 r2 : 0000003c r1 : 380fa100 r0 : c068dda0 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 0005317f Table: 83034000 DAC: 00000015 Process cat (pid: 1042, stack limit = 0xc3ad2270) Stack: (0xc3ad3f38 to 0xc3ad4000) 3f20: c3139180 00000000 3f40: c3bc6500 00001000 be83fb00 c3ad3f80 00001000 c3ad2000 00000000 c0095f3c 3f60: 00000003 c3bc6508 c3bc6500 be83fb00 00000000 00000000 00001000 c0096010 3f80: 00000000 00000000 b6fe2050 00000000 00001000 be83fb00 00000003 00000003 3fa0: c000eb88 c000e9e0 00001000 be83fb00 00000003 be83fb00 00001000 00000000 3fc0: 00001000 be83fb00 00000003 00000003 00000001 00000001 00000000 00000003 3fe0: 000bec8c be83fae0 0000f808 b6ea8d5c 60000010 00000003 7dff7ede 749bedf1 [] (audmux_read_file+0x68/0x2f4) from [] (vfs_read+0xb0/0x144) [] (vfs_read+0xb0/0x144) from [] (sys_read+0x40/0x70) [] (sys_read+0x40/0x70) from [] (ret_fast_syscall+0x0/0x2c) Code: e1a02186 e2822004 e3500000 e7935186 (e7937002) ---[ end trace 4d046e31309023de ]--- Fix the ssi port numbers in sysfs to fix this problem. Reported-by: Joan Carles Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/imx/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 601df80..912a342 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -158,7 +158,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 1; i < 8; i++) { + for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) -- cgit v1.1 From 66bb2a7f835a28a9405f3f6571fbf34156e6bc1e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 5 Apr 2012 10:57:51 -0300 Subject: ASoC: imx-audmux: Check for NULL pointer Check for NULL pointer before accessing it. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/imx/imx-audmux.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 912a342..0fe66c3 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -79,6 +79,9 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + if (!audmux_base) + return -ENOSYS; + if (audmux_clk) clk_prepare_enable(audmux_clk); -- cgit v1.1 From 3fec6b6d5a53d37194735268b9e220f75ca37f19 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 5 Apr 2012 12:28:01 -0600 Subject: ASoC: set idle_bias_off=1 for all platform DAPM contexts The ASoC core currently defaults to using STANDBY rather than OFF for idle ASoC platform devices, which causes a permanent pm_runtime_get() on them. This keeps the device active unnecessarily. This can be especially problematic when the ASoC platform device and DAI device are the same device. The distinction between OFF and STANDBY is likely not relevant for ASoC platform drivers, since they aren't analog devices. So, solve this issue by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this turns out to be a problem, this value could be sourced from the snd_soc_platform_driver, similarly to soc_probe_codec(). Note: Prior to this change, this caused a large (10) runtime_active count for the Tegra I2S controller even when not in use, and a leak in that value as streams were started and stopped. This change probably hides a bug. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a4deebc..8d2ebf5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1087,6 +1087,8 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + platform->dapm.idle_bias_off = 1; + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { -- cgit v1.1 From 8abe05c6eb358967f16bce8a02c88d57c82cfbd6 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 5 Apr 2012 23:11:16 -0600 Subject: ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed the prototype of tegra_i2s_debug_add, but didn't update the dummy inline used when !CONFIG_DEBUG_FS. Fix that. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown Cc: # 3.3 --- sound/soc/tegra/tegra_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 2d98c92..e533499 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -116,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) { } -- cgit v1.1 From 4f32456e5ed4852abc9b555c887dfb3481ea9cab Mon Sep 17 00:00:00 2001 From: Michael Karcher Date: Fri, 6 Apr 2012 15:34:15 +0200 Subject: ALSA: hda - Fix proc output for ADC amp values of CX20549 The CX20549 has only one single input amp on it's input converter widget. Fix printing of values in the codec file in /proc/asound. Signed-off-by: Michael Karcher Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 13 ++++++++++--- sound/pci/hda/patch_conexant.c | 8 ++++---- 3 files changed, 17 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9a9f372..56b4f74 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -851,6 +851,9 @@ struct hda_codec { unsigned int pin_amp_workaround:1; /* pin out-amp takes index * (e.g. Conexant codecs) */ + unsigned int single_adc_amp:1; /* adc in-amp takes no index + * (e.g. CX20549 codec) + */ unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 254ab52..e59e2f0 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -651,9 +651,16 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-In caps: "); print_amp_caps(buffer, codec, nid, HDA_INPUT); snd_iprintf(buffer, " Amp-In vals: "); - print_amp_vals(buffer, codec, nid, HDA_INPUT, - wid_caps & AC_WCAP_STEREO, - wid_type == AC_WID_PIN ? 1 : conn_len); + if (wid_type == AC_WID_PIN || + (codec->single_adc_amp && + wid_type == AC_WID_AUD_IN)) + print_amp_vals(buffer, codec, nid, HDA_INPUT, + wid_caps & AC_WCAP_STEREO, + 1); + else + print_amp_vals(buffer, codec, nid, HDA_INPUT, + wid_caps & AC_WCAP_STEREO, + conn_len); } if (wid_caps & AC_WCAP_OUT_AMP) { snd_iprintf(buffer, " Amp-Out caps: "); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e6eafb1..368617a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -141,7 +141,6 @@ struct conexant_spec { unsigned int hp_laptop:1; unsigned int asus:1; unsigned int pin_eapd_ctrls:1; - unsigned int single_adc_amp:1; unsigned int adc_switching:1; @@ -1111,6 +1110,7 @@ static int patch_cxt5045(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; codec->pin_amp_workaround = 1; + codec->single_adc_amp = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); @@ -4220,7 +4220,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; - if (spec->single_adc_amp) + if (codec->single_adc_amp) idx = 0; return cx_auto_add_volume_idx(codec, label, pfx, cidx, adc_nid, HDA_INPUT, idx); @@ -4275,7 +4275,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) if (cidx < 0) continue; input_conn[i] = spec->imux_info[i].adc; - if (!spec->single_adc_amp) + if (!codec->single_adc_amp) input_conn[i] |= cidx << 8; if (i > 0 && input_conn[i] != input_conn[0]) multi_connection = 1; @@ -4470,7 +4470,7 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: - spec->single_adc_amp = 1; + codec->single_adc_amp = 1; break; case 0x14f15051: add_cx5051_fake_mutes(codec); -- cgit v1.1 From 3edbbb9ec5621478dc3c3b1c66ecb7d177b35c20 Mon Sep 17 00:00:00 2001 From: Michael Karcher Date: Fri, 6 Apr 2012 15:34:16 +0200 Subject: ALSA: hda - Rename capture sources of CX20549 to match common conventions This includes renaming "Line In" to line, also in the mixer settings. Signed-off-by: Michael Karcher Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 368617a..c0a3a17 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -686,27 +686,27 @@ static const struct hda_channel_mode cxt5045_modes[1] = { static const struct hda_input_mux cxt5045_capture_source = { .num_items = 2, .items = { - { "IntMic", 0x1 }, - { "ExtMic", 0x2 }, + { "Internal Mic", 0x1 }, + { "Mic", 0x2 }, } }; static const struct hda_input_mux cxt5045_capture_source_benq = { .num_items = 5, .items = { - { "IntMic", 0x1 }, - { "ExtMic", 0x2 }, - { "LineIn", 0x3 }, - { "CD", 0x4 }, - { "Mixer", 0x0 }, + { "CD", 0x4 }, + { "Internal Mic", 0x1 }, + { "Mic", 0x2 }, + { "Line", 0x3 }, + { "Mixer", 0x0 }, } }; static const struct hda_input_mux cxt5045_capture_source_hp530 = { .num_items = 2, .items = { - { "ExtMic", 0x1 }, - { "IntMic", 0x2 }, + { "Mic", 0x1 }, + { "Internal Mic", 0x2 }, } }; @@ -826,10 +826,10 @@ static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Line Capture Volume", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Line Capture Switch", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), -- cgit v1.1 From cbf2d28e83d47792bd7af000017042dbc59f5df6 Mon Sep 17 00:00:00 2001 From: Michael Karcher Date: Fri, 6 Apr 2012 15:34:17 +0200 Subject: ALSA: hda - fix record volume controls of CX20459 ("Venice") The "input converter" widget of the CX20459 has only one input amplifier, expose that one as "Capture Volume/Capture Switch". The actual record source selection is already exposed through the separately installed input mux. Signed-off-by: Michael Karcher Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 31 ++++++------------------------- 1 file changed, 6 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c0a3a17..4b51c8f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -797,10 +797,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static const struct snd_kcontrol_new cxt5045_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), @@ -821,27 +819,18 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { }; static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { - HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Line Capture Volume", 0x1a, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Line Capture Switch", 0x1a, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), - {} }; static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT), @@ -977,16 +966,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { .put = conexant_mux_enum_put, }, /* Audio input controls */ - HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), { } /* end */ }; -- cgit v1.1 From e6e03daecd2c82437b550ad1a62052c22fdb2b5b Mon Sep 17 00:00:00 2001 From: Michael Karcher Date: Fri, 6 Apr 2012 15:34:18 +0200 Subject: ALSA: hda - Remove CD control from model=benq for CX20549 The ID used for detection of the BenQ R55E actually identifies the Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop series. Schematics on the internet clearly indicate that the "Port C" (analog input connected to record source #4 and mixer input #4) is unconnected. Playing an audio CD through analog playback (using cdplay from cdtools) produces no sound, even with the mixer input labelled "CD" enabled, and the volume control in the CD drive set to maximum. This indicates the connection is really not present. Signed-off-by: Michael Karcher Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4b51c8f..4b36548 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -692,9 +692,8 @@ static const struct hda_input_mux cxt5045_capture_source = { }; static const struct hda_input_mux cxt5045_capture_source_benq = { - .num_items = 5, + .num_items = 4, .items = { - { "CD", 0x4 }, { "Internal Mic", 0x1 }, { "Mic", 0x2 }, { "Line", 0x3 }, @@ -819,9 +818,6 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { }; static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { - HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), -- cgit v1.1 From 51969d62c3b26e887dae734de421b320a296ac58 Mon Sep 17 00:00:00 2001 From: Michael Karcher Date: Fri, 6 Apr 2012 15:34:19 +0200 Subject: ALSA: hda - CX20549 doesn't need pin_amp_workaround. CX20549 (ctx5045) doesn't accept data on index 1 for output pins, as shown in the following hda-var transaction: $ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126 nid = 0x10, verb = 0x300, param = 0xb126 value = 0x0 $ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001 nid = 0x10, verb = 0xb00, param = 0x8001 value = 0x0 Signed-off-by: Michael Karcher Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4b36548..84337e6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1086,7 +1086,6 @@ static int patch_cxt5045(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; - codec->pin_amp_workaround = 1; codec->single_adc_amp = 1; spec->multiout.max_channels = 2; @@ -4443,7 +4442,6 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; - codec->pin_amp_workaround = 1; switch (codec->vendor_id) { case 0x14f15045: @@ -4451,7 +4449,10 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f15051: add_cx5051_fake_mutes(codec); + codec->pin_amp_workaround = 1; break; + default: + codec->pin_amp_workaround = 1; } apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); -- cgit v1.1 From 250f32747e62cb415b85083e247184188f24e566 Mon Sep 17 00:00:00 2001 From: Michael Karcher Date: Fri, 6 Apr 2012 15:34:20 +0200 Subject: ALSA: hda - clean up CX20549 test mixer setup name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls affecting those pins. remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1 really select MIC1, not Mixer out for recording "Mixer out" for recording is not a "pin", adjust comment Signed-off-by: Michael Karcher Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 38 +++++++++++++++++--------------------- 1 file changed, 17 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 84337e6..3848711 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -930,10 +930,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { /* Output controls */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT), /* Modes for retasking pin widgets */ CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), @@ -944,16 +944,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { /* Loopback mixer controls */ - HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -985,10 +985,6 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Start with output sum widgets muted and their output gains at min */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Unmute retasking pin widget output buffers since the default * state appears to be output. As the pin mode is changed by the * user the pin mode control will take care of enabling the pin's @@ -1003,11 +999,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { /* Set ADC connection select to match default mixer setting (mic1 * pin) */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */ -- cgit v1.1 From d3a92d624806a7964ca3122f917ff2ba69e4cdd8 Mon Sep 17 00:00:00 2001 From: Hans Verkuil Date: Sun, 1 Apr 2012 15:24:48 -0300 Subject: [media] Drivers/media/radio: Fix build error On Sunday, April 01, 2012 21:09:34 Tracey Dent wrote: > radio-maxiradio depends on SND_FM801_TEA575X_BOOL to build or will > result in an build error such as: > > Kernel: arch/x86/boot/bzImage is ready (#1) > ERROR: "snd_tea575x_init" [drivers/media/radio/radio-maxiradio.ko] undefined! > ERROR: "snd_tea575x_exit" [drivers/media/radio/radio-maxiradio.ko] undefined! > WARNING: modpost: Found 6 section mismatch(es). > To see full details build your kernel with: > 'make CONFIG_DEBUG_SECTION_MISMATCH=y' > make[1]: *** [__modpost] Error 1 > make: *** [modules] Error 2 > > Select CONFIG_SND_TEA575X to fixes problem and enable > the driver to be built as desired. > > v2: > instead of selecting CONFIG_SND_FM801_TEA575X_BOOL, select > CONFIG_SND_TEA575X, which in turns selects CONFIG_SND_FM801_TEA575X_BOOL > and any other dependencies for it to build. No, this is the correct patch: RADIO_MAXIRADIO should be treated just like RADIO_SF16FMR2, I just didn't realize at the time that it had to be added as a SND_TEA575X dependency. Signed-off-by: Hans Verkuil Tested-by: Shea Levy Acked-by: Mauro Carvalho Chehab Signed-off-by: Mauro Carvalho Chehab --- sound/pci/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 8816804..5ca0939 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -2,8 +2,8 @@ config SND_TEA575X tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 - default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 + depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO + default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO menuconfig SND_PCI bool "PCI sound devices" -- cgit v1.1 From 156d14da4cfc4fe01b705d6e2d22e44c0a2dbecd Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 9 Apr 2012 10:16:32 +0200 Subject: sound: sound/oss/msnd_pinnacle.c: add vfrees At the point of this error-handling code, HAVE_DSPCODEH may be undefined, so free INITCODE and PERMCODE as done elsewhere. A jump and label are introduced to avoid code duplication. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/oss/msnd_pinnacle.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index 2c79d60..536c4c0 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -1294,6 +1294,8 @@ static int __init calibrate_adc(WORD srate) static int upload_dsp_code(void) { + int ret = 0; + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); #ifndef HAVE_DSPCODEH INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE); @@ -1312,7 +1314,8 @@ static int upload_dsp_code(void) memcpy_toio(dev.base, PERMCODE, PERMCODESIZE); if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) { printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); - return -ENODEV; + ret = -ENODEV; + goto out; } #ifdef HAVE_DSPCODEH printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); @@ -1320,12 +1323,13 @@ static int upload_dsp_code(void) printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); #endif +out: #ifndef HAVE_DSPCODEH vfree(INITCODE); vfree(PERMCODE); #endif - return 0; + return ret; } #ifdef MSND_CLASSIC -- cgit v1.1 From 38be95dd3d314bd393a26f6e441ae2c57ef7f064 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Mon, 9 Apr 2012 10:16:35 +0200 Subject: ALSA: sound/isa/sscape.c: add missing resource-release code At the point of this error-handling code, both regions and the dma have been allocated, so free it as done in previous and subsequent error-handling code. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b4a6aa9..8490f59 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1019,13 +1019,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } /* -- cgit v1.1 From fae3d88a5c56c3f836e95c4516da883a48612437 Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Tue, 10 Apr 2012 17:00:35 +0800 Subject: ALSA: hda - hide HDMI/ELD printks unless snd.debug=2 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Also remove two warnings when CONFIG_SND_DEBUG is not set: sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’: sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable] sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable] Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 6 +++--- sound/pci/hda/patch_hdmi.c | 9 ++++----- 2 files changed, 7 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index b58b4b1..4c054f4 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) else buf2[0] = '\0'; - printk(KERN_INFO "HDMI: supports coding type %s:" + _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:" " channels = %d, rates =%s%s\n", cea_audio_coding_type_names[a->format], a->channels, @@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e) { int i; - printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n", + _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - printk(KERN_INFO "HDMI: available speakers:%s\n", buf); + _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 540cd13..83f345f 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -757,8 +757,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) struct hdmi_spec *spec = codec->spec; int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int pin_nid; - int pd = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); int pin_idx; struct hda_jack_tbl *jack; @@ -768,9 +766,10 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) pin_nid = jack->nid; jack->jack_dirty = 1; - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, pd, eldv); + codec->addr, pin_nid, + !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); if (pin_idx < 0) @@ -992,7 +991,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) if (eld->monitor_present) eld_valid = !!(present & AC_PINSENSE_ELDV); - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld_valid); -- cgit v1.1 From 912093bc7c08f59e97faed2c0269e1e5429dcd58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Apr 2012 14:03:41 +0200 Subject: ALSA: hda/realtek - Add a few ALC882 model strings back Since there are still many Acer models that might not be covered by the current fixup table, let's add back a few typical model names so that user can test the fixup without recompiling. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9917e55..e7b2b83 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5399,6 +5399,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { {} }; +static const struct alc_model_fixup alc882_fixup_models[] = { + {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, + {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, + {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {} +}; + /* * BIOS auto configuration */ @@ -5439,7 +5446,8 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, + alc882_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); -- cgit v1.1 From 038d4fef376bc494d4f11072d2ab248414b7d568 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Apr 2012 17:18:12 +0200 Subject: ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co Add GPIO1 setup explicitly for Acer Aspire 493x & co. This could be set by alc_auto_init_amp(), but it's safer to set it more explicitly in the fixup table. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e7b2b83..4eec215 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5269,7 +5269,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x16, 0x99130111 }, /* CLFE speaker */ { 0x17, 0x99130112 }, /* surround speaker */ { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC882_FIXUP_ACER_ASPIRE_8930G] = { .type = ALC_FIXUP_PINS, @@ -5312,7 +5314,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC885_FIXUP_MACPRO_GPIO] = { .type = ALC_FIXUP_FUNC, -- cgit v1.1 From fe97da1f7001ca0f572358462606eb3d1bde3f23 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Apr 2012 08:00:19 +0200 Subject: ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G It's compatible with 8930G. Using the same fixup gives the proper 5.1 sound back. Reported-and-tested-by: Dany Martineau Cc: [v3.3+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4eec215..d25a6f9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5363,6 +5363,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), + SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), -- cgit v1.1 From 29ebe40284c75a5888c601872059fca7e258528d Mon Sep 17 00:00:00 2001 From: Josh Boyer Date: Thu, 12 Apr 2012 13:55:36 -0400 Subject: ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines A user reported that setting model=imac24 used to allow sound to work on their Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models to auto-parser" removed this model option. All Mac machines are now explicitly handled with a quirk and the auto-parser. This adds a quirk for the device found on the Mac Pro 5,1 machines. This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559 [sorted the new entry in the ID number order by tiwai] Reported-by: Gabriel Somlo Signed-off-by: Josh Boyer Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d25a6f9..8f4a484 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5389,6 +5389,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), -- cgit v1.1 From 7e1f7c8a6e517900cd84da1b8ae020f08f286c3b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Apr 2012 17:29:36 +0100 Subject: ASoC: dapm: Ensure power gets managed for line widgets Line widgets had not been included in either the power up or power down sequences so if a widget had an event associated with it that event would never be run. Fix this minimally by adding them to the sequences, we should probably be doing away with the specific widget types as they all have the same priority anyway. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6241490..dc7dbfe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -67,6 +67,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_out_drv] = 10, [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_line] = 10, [snd_soc_dapm_post] = 11, }; @@ -75,6 +76,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_adc] = 1, [snd_soc_dapm_hp] = 2, [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, -- cgit v1.1 From 7d7eb9ea314e992413620610b4d09c9cd5fa8959 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 12 Apr 2012 22:11:25 +0200 Subject: ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace). In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the 'for (;;)' loop, if the 'badness' value returned from fill_and_eval_dacs() is negative, then we'll return from the function without freeing the memory we allocated for 'best_cfg', thus leaking. Fix the leak by kfree()'ing the memory when badness is negative. While I was there I also noticed some trailing whitespace in the function that I removed (along with all other trailing whitespace in the file) - it didn't seem worth-while to do that as two patches, so I hope it's OK that I just did it all as one patch. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8f4a484..2508f81 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3398,8 +3398,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) for (;;) { badness = fill_and_eval_dacs(codec, fill_hardwired, fill_mio_first); - if (badness < 0) + if (badness < 0) { + kfree(best_cfg); return badness; + } debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n", cfg->line_out_type, fill_hardwired, fill_mio_first, badness); @@ -3434,7 +3436,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; fill_hardwired = true; continue; - } + } if (cfg->hp_outs > 0 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { cfg->speaker_outs = cfg->line_outs; @@ -3448,7 +3450,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_HP_OUT; fill_hardwired = true; continue; - } + } break; } @@ -4423,7 +4425,7 @@ static int alc_parse_auto_config(struct hda_codec *codec, static int alc880_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; - static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; + static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids); } @@ -6093,7 +6095,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), @@ -6310,7 +6312,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec, { if (action == ALC_FIXUP_ACT_PRE_PROBE) codec->no_jack_detect = 1; -} +} static const struct alc_fixup alc861_fixups[] = { [ALC861_FIXUP_FSC_AMILO_PI1505] = { @@ -6728,7 +6730,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1), -- cgit v1.1 From 86fc49982369f6918dd9c6eeb70b38ab2303ed0a Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 12 Apr 2012 21:54:34 +0200 Subject: ASoC: cs42l73: don't use negative array index If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using the (negative) return value as array index on the very next line of code - that's bad. Catch the negative return value and propagate it to the caller (which checks for it) and things are a bit more sane :-) Signed-off-by: Jesper Juhl Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3..07c44b7 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -929,6 +929,8 @@ static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) /* MCLKX -> MCLK */ mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / cs42l73_mclkx_coeffs[mclkx_coeff].ratio; -- cgit v1.1 From 8eaeb9393397be8eb700ab38a69c450975463b77 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Apr 2012 11:56:51 +0300 Subject: mfd: Convert twl6040 to i2c driver, and separate it from twl core Complete the separation of the twl6040 from the twl core since it is a separate chip, not part of the twl6030 PMIC. Make the needed Kconfig changes for the depending drivers at the same time to avoid breaking the kernel build (vibra, ASoC components). Signed-off-by: Peter Ujfalusi Reviewed-by: Mark Brown Acked-by: Tony Lindgren Acked-by: Dmitry Torokhov Signed-off-by: Samuel Ortiz --- sound/soc/codecs/Kconfig | 3 +-- sound/soc/codecs/twl6040.c | 3 +-- sound/soc/omap/Kconfig | 2 +- 3 files changed, 3 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6508e8b..59d8efa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,7 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE - select SND_SOC_TWL6040 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE @@ -276,7 +276,6 @@ config SND_SOC_TWL4030 tristate config SND_SOC_TWL6040 - select TWL6040_CORE tristate config SND_SOC_UDA134X diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2d8c6b8..dc7509b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include @@ -1528,7 +1527,7 @@ static int twl6040_resume(struct snd_soc_codec *codec) static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; - struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); + struct twl6040_codec_data *pdata = dev_get_platdata(codec->dev); struct platform_device *pdev = container_of(codec->dev, struct platform_device, dev); int ret = 0; diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e00dd0b..deafbfa 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -97,7 +97,7 @@ config SND_OMAP_SOC_SDP3430 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 -- cgit v1.1 From a7dbb603423d499acacefb5fad65d2b406f16370 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 17 Apr 2012 18:00:11 +0100 Subject: ASoC: core: Fix card RTD count for deferred probe. Currently we increment the number of RTD's per card during the DAI link bind. This can cause an incorrect RTD count when we cannot find a component and defer the probe (and hence perform the DAI link bind for the card again). Fix the count so that it is cleared before every card registration and bind attempt. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8d2ebf5..3a4e93e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3119,6 +3119,7 @@ int snd_soc_register_card(struct snd_soc_card *card) GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; + card->num_rtd = 0; card->rtd_aux = &card->rtd[card->num_links]; for (i = 0; i < card->num_links; i++) -- cgit v1.1 From f2ec52d4c3698c995c89c579c34d818eab589d8b Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Tue, 17 Apr 2012 17:03:42 -0700 Subject: ALSA: fix core/vmaster.c kernel-doc warning Fix kernel-doc warning in sound/core/vmaster.c: Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 14a286a..8575861 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -419,6 +419,7 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master); * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control * @kcontrol: vmaster kctl element * @hook: the hook function + * @private_data: the private_data pointer to be saved * * Adds the given hook to the vmaster control element so that it's called * at each time when the value is changed. -- cgit v1.1 From cdf27f373781d8740b874b0b5c18142df32ebb52 Mon Sep 17 00:00:00 2001 From: Paul Mundt Date: Tue, 17 Apr 2012 19:13:04 -0700 Subject: ASoC: fsi: update for dmaengine prep_slave_sg fallout. Leading up to the ->device_prep_slave_sg change in 185ecb5f4fd43911c35956d4cc7d94a1da30417f 'dmaengine: add context parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was added in place to guard against the API change, though the fsi driver wasn't updated in the process (presumably its dmaengine support hadn't been merged yet at the time). This trivially switches over to the new wrapper and gets it building again. Signed-off-by: Paul Mundt Acked-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 378cc5b..74ed2df 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1001,11 +1001,10 @@ static void fsi_dma_do_tasklet(unsigned long data) sg_dma_address(&sg) = buf; sg_dma_len(&sg) = len; - desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | - DMA_CTRL_ACK); + desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { - dev_err(dai->dev, "device_prep_slave_sg() fail\n"); + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; } -- cgit v1.1 From 118cb4a408e1c4021ac85d6c05da66bb6f57e556 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 07:33:27 +0200 Subject: ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1 Through the transition to the auto-parser, the support for Quanta/Gericom KN1 got broken. There are two problems behind it: - This machine doesn't like the default COEF setup for ALC260 we take now as default - BIOS doesn't set the pins correctly at all; especially the machine uses only the pin 0x0f for both headphone and speaker This patch adds the fixup as a workaround for these issues. Reported-and-tested-by: Uros Vampl Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 49 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 45 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2508f81..e65e354 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1445,6 +1445,13 @@ enum { ALC_FIXUP_ACT_BUILD, }; +static void alc_apply_pincfgs(struct hda_codec *codec, + const struct alc_pincfg *cfg) +{ + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} + static void alc_apply_fixup(struct hda_codec *codec, int action) { struct alc_spec *spec = codec->spec; @@ -1478,9 +1485,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) snd_printdd(KERN_INFO "hda_codec: %s: " "Apply pincfg for %s\n", codec->chip_name, modelname); - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, - cfg->val); + alc_apply_pincfgs(codec, cfg); break; case ALC_FIXUP_VERBS: if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) @@ -4861,6 +4866,7 @@ enum { ALC260_FIXUP_GPIO1_TOGGLE, ALC260_FIXUP_REPLACER, ALC260_FIXUP_HP_B1900, + ALC260_FIXUP_KN1, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -4888,6 +4894,36 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, } } +static void alc260_fixup_kn1(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct alc_pincfg pincfgs[] = { + { 0x0f, 0x02214000 }, /* HP/speaker */ + { 0x12, 0x90a60160 }, /* int mic */ + { 0x13, 0x02a19000 }, /* ext mic */ + { 0x18, 0x01446000 }, /* SPDIF out */ + /* disable bogus I/O pins */ + { 0x10, 0x411111f0 }, + { 0x11, 0x411111f0 }, + { 0x14, 0x411111f0 }, + { 0x15, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { } + }; + + switch (action) { + case ALC_FIXUP_ACT_PRE_PROBE: + alc_apply_pincfgs(codec, pincfgs); + break; + case ALC_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_NONE; + break; + } +} + static const struct alc_fixup alc260_fixups[] = { [ALC260_FIXUP_HP_DC5750] = { .type = ALC_FIXUP_PINS, @@ -4938,7 +4974,11 @@ static const struct alc_fixup alc260_fixups[] = { .v.func = alc260_fixup_gpio1_toggle, .chained = true, .chain_id = ALC260_FIXUP_COEF, - } + }, + [ALC260_FIXUP_KN1] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc260_fixup_kn1, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -4948,6 +4988,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), + SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} -- cgit v1.1 From 3e843196c697ee2c319d96e861980fb4c3e04e24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 12:04:03 +0200 Subject: ALSA: hda/sigmatel - Fix inverted mute LED While refactoring the mute-LED handling for HP laptops, I messed up the polarity check in a wrong way. The red (or the mute-LED if any) should appear in the muted state, corresponding to GPIO on. Reported-by: Mikko Vinni Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 33a9946..4742cac 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5063,12 +5063,11 @@ static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) if (spec->gpio_led_polarity) muted = !muted; - /*polarity defines *not* muted state level*/ if (!spec->vref_mute_led_nid) { if (muted) - spec->gpio_data &= ~spec->gpio_led; /* orange */ + spec->gpio_data |= spec->gpio_led; else - spec->gpio_data |= spec->gpio_led; /* white */ + spec->gpio_data &= ~spec->gpio_led; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } else { -- cgit v1.1 From 590b4775d6b628c7ad215fd0335a0a787032e2dd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 19 Apr 2012 00:00:27 -0700 Subject: ALSA: workaround: change the timing of alsa_sound_last_init() Current alsa_sound_last_init() was called as __initcall(). So, on current ALSA, only devices that had been properly registered at this point were shown. So, it will show "No soundcards found" if driver requests probe deferment. it's often misleading. This patch delays the timing of alsa_sound_last_init() as workaround. Signed-off-by: Kuninori Morimoto Reviwed-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/last.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/last.c b/sound/last.c index bdd0857..7ffc182 100644 --- a/sound/last.c +++ b/sound/last.c @@ -38,4 +38,4 @@ static int __init alsa_sound_last_init(void) return 0; } -__initcall(alsa_sound_last_init); +late_initcall_sync(alsa_sound_last_init); -- cgit v1.1 From ca3649de026ff95c6f2847e8d096cf2f411c02b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 15:15:25 +0200 Subject: ALSA: hda/conexant - Don't set HP pin-control bit unconditionally Some output pins on Conexant chips have no HP control bit, but the auto-parser initializes these pins unconditionally with PIN_HP. Check the pin-capability and avoid the HP bit if not supported. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d29d6d3..f52c9ef 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3951,9 +3951,14 @@ static void cx_auto_init_output(struct hda_codec *codec) int i; mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids); - for (i = 0; i < cfg->hp_outs; i++) + for (i = 0; i < cfg->hp_outs; i++) { + unsigned int val = PIN_OUT; + if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) & + AC_PINCAP_HP_DRV) + val |= AC_PINCTL_HP_EN; snd_hda_codec_write(codec, cfg->hp_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + } mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins); -- cgit v1.1 From d70f363222ef373c2037412f09a600357cfa1c7a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Apr 2012 15:18:08 +0200 Subject: ALSA: hda/conexant - Set up the missing docking-station pins ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the docking-station ports, but BIOS doesn't initialize for these pins. Thus, like the former X200, we need to set up the pins manually in the driver. The odd part is that the same PCI SSID is used for X200 and T400, thus we need to prepare individual fixup tables for cx5051 and others. Bugzilla entries: https://bugzilla.redhat.com/show_bug.cgi?id=808559 https://bugzilla.redhat.com/show_bug.cgi?id=806217 https://bugzilla.redhat.com/show_bug.cgi?id=810697 Reported-by: Josh Boyer Reported-by: Jens Taprogge Tested-by: Jens Taprogge Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 28 +++++++++++++++++++++++++--- 1 file changed, 25 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f52c9ef..58b5de4 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4367,8 +4367,10 @@ static void apply_pin_fixup(struct hda_codec *codec, enum { CXT_PINCFG_LENOVO_X200, + CXT_PINCFG_LENOVO_TP410, }; +/* ThinkPad X200 & co with cxt5051 */ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4376,15 +4378,33 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { {} }; +/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */ +static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = { + { 0x19, 0x042110ff }, /* HP (seq# overridden) */ + { 0x1a, 0x21a190f0 }, /* dock-mic */ + { 0x1c, 0x212140ff }, /* dock-HP */ + {} +}; + static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410, }; -static const struct snd_pci_quirk cxt_fixups[] = { +static const struct snd_pci_quirk cxt5051_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; +static const struct snd_pci_quirk cxt5066_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + {} +}; + /* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches * can be created (bko#42825) */ @@ -4421,11 +4441,13 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f15051: add_cx5051_fake_mutes(codec); + apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl); + break; + default: + apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl); break; } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); - err = cx_auto_search_adcs(codec); if (err < 0) return err; -- cgit v1.1 From 5ac57550f279c3d991ef0b398681bcaca18169f7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 20 Apr 2012 10:01:46 +0200 Subject: ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E According to the reporter, external mic starts to work if the laptop-dmic model is used. According to BIOS pin config, all pins are consistent with the alc269vb_laptop_dmic fixup, except for the external mic, which is not present. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/950490 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e65e354..818f90b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6109,6 +6109,7 @@ static const struct alc_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- cgit v1.1 From 1a38336b8611a04f0a624330c1f815421f4bf5f4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 12 Apr 2012 19:47:11 +0100 Subject: ASoC: wm8994: Improve sequencing of AIF channel enables This ensures a clean startup of the channels, without this change some use cases could result in issues in a small proportion of cases. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 276 +++++++++++++++++++++++++++++++++++++--------- 1 file changed, 222 insertions(+), 54 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642..6c1fe3a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1000,61 +1000,170 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } -static int late_enable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; + int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; + int dac; + int adc; + int val; + + switch (control->type) { + case WM8994: + case WM8958: + mask |= WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA; + break; + default: + break; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (wm8994->aif1clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, - WM8994_AIF1CLK_ENA); - wm8994->aif1clk_enable = 0; - } - if (wm8994->aif2clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, - WM8994_AIF2CLK_ENA); - wm8994->aif2clk_enable = 0; - } + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); + if ((val & WM8994_AIF1ADCL_SRC) && + (val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA; + else if (!(val & WM8994_AIF1ADCL_SRC) && + !(val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + else + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_2); + if ((val & WM8994_AIF1DACL_SRC) && + (val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA; + else if (!(val & WM8994_AIF1DACL_SRC) && + !(val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + else + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, mask, + WM8994_AIF1ADC1R_ENA | + WM8994_AIF1ADC1L_ENA | + WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC2L_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, mask, + WM8994_AIF1DAC1R_ENA | + WM8994_AIF1DAC1L_ENA | + WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC2L_ENA); break; - } - /* We may also have postponed startup of DSP, handle that. */ - wm8958_aif_ev(w, kcontrol, event); + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF2DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF1DSPCLK_ENA, val); + break; + } return 0; } -static int late_disable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int dac; + int adc; + int val; switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_1); + if ((val & WM8994_AIF2ADCL_SRC) && + (val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCR_ENA; + else if (!(val & WM8994_AIF2ADCL_SRC) && + !(val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCL_ENA; + else + adc = WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA; + + + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_2); + if ((val & WM8994_AIF2DACL_SRC) && + (val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACR_ENA; + else if (!(val & WM8994_AIF2DACL_SRC) && + !(val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACL_ENA; + else + dac = WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA); + break; + + case SND_SOC_DAPM_PRE_PMD: case SND_SOC_DAPM_POST_PMD: - if (wm8994->aif1clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, 0); - wm8994->aif1clk_disable = 0; - } - if (wm8994->aif2clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, 0); - wm8994->aif2clk_disable = 0; - } + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF1DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF2DSPCLK_ENA, val); break; } return 0; } -static int aif1clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1071,8 +1180,8 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, return 0; } -static int aif2clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1089,6 +1198,63 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) { + aif1clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + aif2clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + wm8994->aif2clk_enable = 0; + } + break; + } + + /* We may also have postponed startup of DSP, handle that. */ + wm8958_aif_ev(w, kcontrol, event); + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + aif1clk_ev(w, kcontrol, event); + wm8994->aif1clk_disable = 0; + } + if (wm8994->aif2clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + aif2clk_ev(w, kcontrol, event); + wm8994->aif2clk_disable = 0; + } + break; + } + + return 0; +} + static int adc_mux_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1385,9 +1551,9 @@ static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, @@ -1416,8 +1582,10 @@ SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1470,30 +1638,30 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPINTCLK", SND_SOC_NOPM, 1, 0, NULL, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 9, 0), + 0, SND_SOC_NOPM, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 8, 0), + 0, SND_SOC_NOPM, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, + SND_SOC_NOPM, 9, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, + SND_SOC_NOPM, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 11, 0), + 0, SND_SOC_NOPM, 11, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 10, 0), + 0, SND_SOC_NOPM, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, + SND_SOC_NOPM, 11, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev, + SND_SOC_NOPM, 10, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, @@ -1520,14 +1688,14 @@ SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, dac1r_mix, ARRAY_SIZE(dac1r_mix)), SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 13, 0), + SND_SOC_NOPM, 13, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 12, 0), + SND_SOC_NOPM, 12, 0), SND_SOC_DAPM_AIF_IN_E("AIF2DACL", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 13, 0, wm8958_aif_ev, + SND_SOC_NOPM, 13, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 12, 0, wm8958_aif_ev, + SND_SOC_NOPM, 12, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), -- cgit v1.1 From de050acaa1fdba4852cb195baf2bfed75368e0be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Apr 2012 20:28:10 +0100 Subject: ASoC: wm_hubs: Make sure we don't disable differential line outputs While we need to clean up unused single ended line outputs we don't want to do this if the outputs are in differential mode. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f13f288..6c028c4 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); - int val; + int mask, val; switch (level) { case SND_SOC_BIAS_STANDBY: @@ -1047,6 +1047,13 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: /* Turn off any unneded single ended outputs */ val = 0; + mask = 0; + + if (hubs->lineout1_se) + mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; if (hubs->lineout1_se && hubs->lineout1n_ena) val |= WM8993_LINEOUT1N_ENA; @@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, val |= WM8993_LINEOUT2P_ENA; snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, - WM8993_LINEOUT1N_ENA | - WM8993_LINEOUT1P_ENA | - WM8993_LINEOUT2N_ENA | - WM8993_LINEOUT2P_ENA, - val); + mask, val); /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, -- cgit v1.1 From c34ce320d9fe328e3272def20b152f39ccfa045e Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Tue, 24 Apr 2012 15:24:43 +0800 Subject: ASoC: core: check of_property_count_strings failure Signed-off-by: Richard Zhao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4e93e..b390f00 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3631,10 +3631,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, int i, ret; num_routes = of_property_count_strings(np, propname); - if (num_routes & 1) { + if (num_routes < 0 || num_routes & 1) { dev_err(card->dev, - "Property '%s's length is not even\n", - propname); + "Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; -- cgit v1.1 From a3a53fe1545a87337cc539f415810128bbdad465 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 25 Apr 2012 11:29:47 +0200 Subject: ASoC: bf5xx-ssm2602: Set DAI format Commit 980b0bc69 ("ASoC: blackfin: Use dai_fmt") converted the blackfin ASoC machine drivers to use the dai_links dai_fmt field to setup their DAI format. For the bf5xx-ssm2602 the commit removed the manual call to snd_soc_dai_set_fmt, but missed to set the dai_links dai_fmt field. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ssm2602.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index df3ac73..b39ad35 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { .name = "ssm2602", @@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; -- cgit v1.1 From e875c1e3e758447ba81ca450d89434b3b0496d37 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Eric=20B=C3=A9nard?= Date: Sun, 29 Apr 2012 17:37:57 +0200 Subject: ASoC: tlv312aic23: unbreak resume MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit * commit f9dfbf9 "ASoC: tlv320aic23: convert to soc-cache" leads to a bug preventing resumeof the codec as regmap expects a 9 bits data register but 0xFFFF is passed in tlv320aic23_set_bias_level and this values gets cached preventing any write to the TLV320AIC23_PWR register as the final value produced by regmap is (register << 9) | value * this patch solves the problem by only working on the 9 bits the register contains. Signed-off-by: Eric Bénard Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/tlv320aic23.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f9..df1e07f 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; switch (level) { case SND_SOC_BIAS_ON: @@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); - snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff); break; } codec->dapm.bias_level = level; -- cgit v1.1 From 30facd4d51d630b6cba386badd7f42456962089b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Apr 2012 20:11:55 +0100 Subject: ASoC: wm8350: Don't use locally allocated codec struct The core allocates the live copies, we shouldn't try to duplicate it and were buggy trying to do so as we were using uninitialised data for the control data. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 8c4c959..aa12c6b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -60,7 +60,7 @@ struct wm8350_jack_data { }; struct wm8350_data { - struct snd_soc_codec codec; + struct wm8350 *wm8350; struct wm8350_output out1; struct wm8350_output out2; struct wm8350_jack_data hpl; @@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) { - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report; @@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work) static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_jack_data *jack = NULL; switch (irq - wm8350->irq_base) { @@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); static irqreturn_t wm8350_mic_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report = 0; @@ -1536,6 +1536,8 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->wm8350 = wm8350; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) priv->supplies[i].supply = supply_names[i]; @@ -1544,7 +1546,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - wm8350->codec.codec = codec; codec->control_data = wm8350; /* Put the codec into reset if it wasn't already */ -- cgit v1.1 From 06412088ce98f745405b8f65cfc51ddd6b842bbf Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Heiko=20St=C3=BCbner?= Date: Mon, 30 Apr 2012 13:17:21 +0200 Subject: ASoC: s3c2412-i2s: Fix dai registration As s3c2412-i2s is using the s3c_i2sv2 it should call the more specialised s3c_i2sv2_register_dai instead of simply calling snd_soc_register_dai. Without this call the snd_soc_dai_ops structure isn't initialised correctly. Signed-off-by: Heiko Stuebner Signed-off-by: Mark Brown --- sound/soc/samsung/s3c2412-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 7218507..79fbeea 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); } static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) -- cgit v1.1 From fad9365bcc2a69ae16adc092e8ac192354980665 Mon Sep 17 00:00:00 2001 From: Oleg Matcovschi Date: Tue, 24 Apr 2012 19:02:02 -0700 Subject: ASoC: omap-pcm: Free dma buffers in case of error. Signed-off-by: Oleg Matcovschi Acked-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a59bd35..5a649da 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) } out: + /* free preallocated buffers in case of error */ + if (ret) + omap_pcm_free_dma_buffers(pcm); + return ret; } -- cgit v1.1 From c914f55f7cdfafe9d7d5b248751902c7ab57691e Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Mon, 30 Apr 2012 19:39:22 +0100 Subject: ALSA: echoaudio: Remove incorrect part of assertion This assertion seems to imply that chip->dsp_code_to_load is a pointer. It's actually an integer handle on the actual firmware, and 0 has no special meaning. The assertion prevents initialisation of a Darla20 card, but would also affect other models. It seems it was introduced in commit dd7b254d. ALSA sound/pci/echoaudio/echoaudio.c:2061 Echoaudio driver starting... ALSA sound/pci/echoaudio/echoaudio.c:1969 chip=ebe4e000 ALSA sound/pci/echoaudio/echoaudio.c:2007 pci=ed568000 irq=19 subdev=0010 Init hardware... ALSA sound/pci/echoaudio/darla20_dsp.c:36 init_hw() - Darla20 ------------[ cut here ]------------ WARNING: at sound/pci/echoaudio/echoaudio_dsp.c:478 init_hw+0x1d1/0x86c [snd_darla20]() Hardware name: Dell DM051 BUG? (!chip->dsp_code_to_load || !chip->comm_page) Signed-off-by: Mark Hills Cc: Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio_dsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 64417a7..d8c670c 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -475,7 +475,7 @@ static int load_firmware(struct echoaudio *chip) const struct firmware *fw; int box_type, err; - if (snd_BUG_ON(!chip->dsp_code_to_load || !chip->comm_page)) + if (snd_BUG_ON(!chip->comm_page)) return -EPERM; /* See if the ASIC is present and working - only if the DSP is already loaded */ -- cgit v1.1 From f5c53d898cc34079373c63a290528963db31d681 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 10:07:33 +0200 Subject: ALSA: hda/realtek - Add a fixup for Acer Aspire 5739G Acer Aspire 5739G requires the same fix-up for 4930G to support the surround / bass speakers. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 818f90b..27d0f63 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5405,6 +5405,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G", + ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), -- cgit v1.1 From bca40138558f0b39357fd1ca477868e4f52f4b1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 11:13:14 +0200 Subject: ALSA: hda/realtek - Add missing CD-input pin for MSI-7350 mobo Reported-by: Philipp Matthias Hahn Cc: [v3.3+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 27d0f63..8ea613e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5440,6 +5440,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- cgit v1.1 From 42eb92380f73f28e3a5a51973af1183fdbac82f2 Mon Sep 17 00:00:00 2001 From: Andre Schramm Date: Mon, 7 May 2012 18:52:51 +0200 Subject: ALSA: hdsp - Provide ioctl_compat snd_hdsp uses its own ioctls to acquire config- and status information. Expose the corresponding ioctl handler via ioctl_compat, so that 32bit applications can use it on 64bit kernels. Signed-off-by: Andre Schramm Reviewed-by: Adrian Knoth Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b68cdec..0b2aea2 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5170,6 +5170,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) strcpy(hw->name, "HDSP hwdep interface"); hw->ops.ioctl = snd_hdsp_hwdep_ioctl; + hw->ops.ioctl_compat = snd_hdsp_hwdep_ioctl; return 0; } -- cgit v1.1 From af741c150f66db8d1da6f82ac75e2571f7f1dd38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 7 May 2012 18:09:48 +0200 Subject: ALSA: hda/realtek - Call alc_auto_parse_customize_define() always after fixup The call for alc_auto_parse_customize_define() must be done after the fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work properly (e.g. HP RP5800 with ALC662 codec). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8ea613e..7810913 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5641,13 +5641,13 @@ static int patch_alc262(struct hda_codec *codec) snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80); } #endif - alc_auto_parse_customize_define(codec); - alc_fix_pll_init(codec, 0x20, 0x0a, 10); alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc262_parse_auto_config(codec); if (err < 0) @@ -6252,8 +6252,6 @@ static int patch_alc269(struct hda_codec *codec) spec->mixer_nid = 0x0b; - alc_auto_parse_customize_define(codec); - err = alc_codec_rename_from_preset(codec); if (err < 0) goto error; @@ -6286,6 +6284,8 @@ static int patch_alc269(struct hda_codec *codec) alc269_fixup_tbl, alc269_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); if (err < 0) @@ -6862,8 +6862,6 @@ static int patch_alc662(struct hda_codec *codec) /* handle multiple HPs as is */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; - alc_auto_parse_customize_define(codec); - alc_fix_pll_init(codec, 0x20, 0x04, 15); err = alc_codec_rename_from_preset(codec); @@ -6880,6 +6878,9 @@ static int patch_alc662(struct hda_codec *codec) alc_pick_fixup(codec, alc662_fixup_models, alc662_fixup_tbl, alc662_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); + + alc_auto_parse_customize_define(codec); + /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) -- cgit v1.1 From 619a341b78f17fb86d92e89c04612676cd05e26f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 8 May 2012 16:30:59 +0200 Subject: Revert "ALSA: hda - Set codec to D3 forcibly even if not used" This reverts commit 785f857d1cb0856b612b46a0545b74aa2596e44a. The commit causes a problem with the wrong D3 state after suspend because the call of hda_set_power_state() involves with the power-up sequence, which changes the power_count, and this confuses the resume sequence that checks the power_count as well. Originally, this go-to-D3 sequence should be a simple task without the power-up sequence. But, it'd need some proper sanity checks in the case of power-saved state, so it's not too easy to write now in the 3.4-rc cycle. In short, the safest option now is to revert this affecting commit. Of course, we need to clean up and robustify the power-saving code better for 3.5 kernel. Reported-by: Konstantin Khlebnikov Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ---- sound/pci/hda/hda_intel.c | 14 +++++++++++++- 2 files changed, 13 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7a8fcc4..841475c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5444,10 +5444,6 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); - else /* forcibly change the power to D3 even if not used */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D3); if (codec->patch_ops.post_suspend) codec->patch_ops.post_suspend(codec); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c19e71a..6e958bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2351,6 +2351,17 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ +static int snd_hda_codecs_inuse(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + if (snd_hda_codec_needs_resume(codec)) + return 1; + } + return 0; +} + static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); @@ -2397,7 +2408,8 @@ static int azx_resume(struct pci_dev *pci) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + if (snd_hda_codecs_inuse(chip->bus)) + azx_init_chip(chip, 1); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); -- cgit v1.1 From 32cf4023e689ad5b3a81a749d8cc99d7f184cb99 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 May 2012 11:05:55 +0200 Subject: ALSA: HDA: Lessen CPU usage when waiting for chip to respond When an IRQ for some reason gets lost, we wait up to a second using udelay, which is CPU intensive. This patch improves the situation by waiting about 30 ms in the CPU intensive mode, then stepping down to using msleep(2) instead. In essence, we trade some granularity in exchange for less CPU consumption when the waiting time is a bit longer. As a result, PulseAudio should no longer be killed by the kernel for taking up to much RT-prio CPU time. At least not for *this* reason. Signed-off-by: David Henningsson Tested-by: Arun Raghavan Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e958bf..1f35052 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -783,11 +783,13 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, { struct azx *chip = bus->private_data; unsigned long timeout; + unsigned long loopcounter; int do_poll = 0; again: timeout = jiffies + msecs_to_jiffies(1000); - for (;;) { + + for (loopcounter = 0;; loopcounter++) { if (chip->polling_mode || do_poll) { spin_lock_irq(&chip->reg_lock); azx_update_rirb(chip); @@ -803,7 +805,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } if (time_after(jiffies, timeout)) break; - if (bus->needs_damn_long_delay) + if (bus->needs_damn_long_delay || loopcounter > 3000) msleep(2); /* temporary workaround */ else { udelay(10); -- cgit v1.1 From c8587193ba511b788a9888e5e701a9747e70c0d8 Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Wed, 9 May 2012 12:57:05 +0200 Subject: ASoC: sh: fix migor.c compilation Fix a recent compilation breakage, caused by a change in SH clock API. Signed-off-by: Guennadi Liakhovetski Signed-off-by: Mark Brown --- sound/soc/sh/migor.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 9d9ad8d..8526e1e 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -35,7 +35,7 @@ static unsigned long siumckb_recalc(struct clk *clk) return codec_freq; } -static struct clk_ops siumckb_clk_ops = { +static struct sh_clk_ops siumckb_clk_ops = { .recalc = siumckb_recalc, }; -- cgit v1.1 From 5807c3bf68eb489032ca8ff70b3d3c833fd8172b Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Fri, 11 May 2012 12:54:45 -0500 Subject: ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB. Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l73.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 07c44b7..3686417 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -568,22 +568,22 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { attn_tlv), SOC_SINGLE_TLV("SPK-IP Mono Volume", - CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-XSP Mono Volume", - CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-ASP Mono Volume", - CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-VSP Mono Volume", - CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-IP Mono Volume", - CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-XSP Mono Volume", - CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-ASP Mono Volume", - CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-VSP Mono Volume", - CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), -- cgit v1.1 From b0791dda813c179e539b0fc1ecd3f5f30f2571e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 15 May 2012 08:07:31 +0200 Subject: ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops BIOS on some HP laptops don't set the speaker-pins as fixed but expose as jacks, and this confuses the driver as if these pins are jack-detectable. As a result, the machine doesn't get sounds from speakers because the driver prepares the power-map update via jack unsol events which never come up in reality. The bug was introduced in some time in 3.2 for enabling the power-mapping feature. This patch fixes the problem by replacing the check of the persistent power-map bits with a proper is_jack_detectable() call. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43240 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4742cac..2cb1e08 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4415,9 +4415,9 @@ static int stac92xx_init(struct hda_codec *codec) def_conf = get_defcfg_connect(def_conf); /* skip any ports that don't have jacks since presence * detection is useless */ - if (def_conf != AC_JACK_PORT_COMPLEX) { - if (def_conf != AC_JACK_PORT_NONE) - stac_toggle_power_map(codec, nid, 1); + if (def_conf != AC_JACK_PORT_NONE && + !is_jack_detectable(codec, nid)) { + stac_toggle_power_map(codec, nid, 1); continue; } if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) { -- cgit v1.1 From c7f5f2389377b66028bc129890aa653deafe8d39 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 15 May 2012 18:13:00 +0100 Subject: ASoC: wm8994: Fix AIF2ADC power down Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6c1fe3a..2de12eb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1144,7 +1144,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA, 0); - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA, 0); -- cgit v1.1