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

[alsa-devel] [PATCH v4 01/19] ALSA: Oxygen: Add the separate SPI waiting function
by Roman Volkov 30 Jan '14

30 Jan '14

The oxygen_wait_spi() function now performs waiting when the SPI bus completes a transaction. Introduce the timeout error checking and increase timeout to 200 from 40. Signed-off-by: Roman Volkov <v1ron(a)mail.ru> --- sound/pci/oxygen/oxygen_io.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 521eae4..471e64a 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -194,6 +194,21 @@ void oxygen_write_ac97_masked(struct oxygen *chip, unsigned int codec, } EXPORT_SYMBOL(oxygen_write_ac97_masked); +static int oxygen_wait_spi(struct oxygen *chip) +{ + /* Higher timeout to be sure: 200 us, + Actual transaction should not need more than 40 us */ + unsigned int count; + for (count = 50; count > 0; count--) { + udelay(4); + if ((oxygen_read8(chip, OXYGEN_SPI_CONTROL) & + OXYGEN_SPI_BUSY) == 0) + return 0; + } + snd_printk(KERN_ERR "Oxygen: SPI wait timeout\n"); + return -EIO; +} + void oxygen_write_spi(struct oxygen *chip, u8 control, unsigned int data) { unsigned int count; -- 1.7.10.4

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[alsa-devel] [PATCH 1/2] ASoC: Samsung: Remove invalid dependencies
by Sachin Kamat 30 Jan '14

30 Jan '14

These symbols got eliminated when non-DT support for Exynos was removed. Remove them. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1f369c69469c..c006d4aafd5a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -145,7 +145,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirks for another Dell laptop
by Hui Wang 30 Jan '14

30 Jan '14

When we plug a 3-ring headset on the Dell machine (Vendor ID: 0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson <david.henningsson(a)canonical.com> Tested-by: Doro Wu <fan-cheng.wu(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d723c1f..56a8f18 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4294,6 +4294,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x064d, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ASoC: fsl: imx-sgtl5000: No need to initialize local variable data
by Philipp Zabel 30 Jan '14

30 Jan '14

The first access to this variable is an assignment to the result of a kzalloc function call. Remove the superfluous initialisation. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- sound/soc/fsl/imx-sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae7..3e60b4c 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data = NULL; + struct imx_sgtl5000_data *data; int int_port, ext_port; int ret; -- 1.8.5.3

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[alsa-devel] Regression due to 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP"
by Stephen Warren 30 Jan '14

30 Jan '14

Takashi, It looks like commit 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP" has caused a regression on NVIDIA graphics cards. It's been reported to us by at least two separate users. On NVIDIA GPUs, it appears that if multiple pin complex widgets have their input mux set to the same audio output widget and PIN_OUT, then only one of the pin complex widgets will actually receive the audio. That's the reason we need to dynamically toggle PIN_OUT on and off, so that only actually active pin complex widgets are enabled. Without this, audio will be routed to the same pin complex all the time, rather than the user-selected device. I'm not sure if this is the way HDA CODECs are meant to work, but it certainly is how they do work:-) So, I'd like to basically revert 6169b673618b "ALSA: hda - Always turn on pins for HDMI/DP", and return the code to the state it was after commit 384a48d71520 "ALSA: hda: HDMI: Support codecs with fewer cvts than pins". However, I obviously don't want to re-introduce the regression that your commit solved. I think we should solve this by including the code for both behaviours in the kernel, and having either a flag in struct hdmi_spec, or some extra ops in struct hdmi_ops, select between the two paths. Does that seem reasonable? If so, the only remaining question is: which behaviour should be default a/k/a should I blacklist the problematic GM965 (mentioned in https://bugzilla.kernel.org/show_bug.cgi?id=51421) against having the dynamic PIN_OUT behaviour, or whitelist NVIDIA GPUs as requiring the dynamic PIN_OUT behaviour? I know that my patch worked fine on my wife's Intel chipset, but I have no idea if the dynamic PIN_OUT behaviour was actually needed there, or simply tolerated, and I have no way to test much aside from NVIDIA GPUs. That all said, the dynamic PIN_OUT behaviour seems to make most sense to me, or some systems may end up with audio being sent to multiple outputs when only one was expected.

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[alsa-devel] [PATCH v2 1/2] ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
by Philipp Zabel 30 Jan '14

30 Jan '14

In asynchronous mode, a RxFS and RxClk connection needs to be made between two ports. Add a define for the bit to be set in the *SEL fields. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 50b09e9..63b48f8 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -24,6 +24,10 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +/* TFSEL/TCSEL/RFSEL/RCSEL can be sourced from Rx/Tx clocks on i.MX51/53/6Q */ +#define IMX_AUDMUX_RXFS 0x8 +#define IMX_AUDMUX_RXCLK 0x8 + /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ #define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) #define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -- 1.8.5.3

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[alsa-devel] [PATCH 1/2] ASoC: imx-audmux: add RXFS/RXCLK defines for 6-wire connections
by Philipp Zabel 30 Jan '14

30 Jan '14

In asynchronous mode, a RxFS and RxClk connection needs to be made between two ports. Add a define for the bit to be set in the *SEL fields. Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h index 50b09e9..63b48f8 100644 --- a/include/dt-bindings/sound/fsl-imx-audmux.h +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -24,6 +24,10 @@ #define MX51_AUDMUX_PORT6 5 #define MX51_AUDMUX_PORT7 6 +/* TFSEL/TCSEL/RFSEL/RCSEL can be sourced from Rx/Tx clocks on i.MX51/53/6Q */ +#define IMX_AUDMUX_RXFS 0x8 +#define IMX_AUDMUX_RXCLK 0x8 + /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ #define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) #define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -- 1.8.5.3

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[alsa-devel] [PATCH 0/2] ASoC: core: volume control using signed register values
by Markus Pargmann 29 Jan '14

29 Jan '14

Hi, There are a lot of codecs using negative and positive volume control ranges as values in their registers. These two patches add support for arbitrary size signed integer values in registers. Regards, Markus Markus Pargmann (2): ASoC: core: Add signed register volume control logic ASoC: soc.h: Add SOC_DOUBLE_R_S_TLV include/sound/soc.h | 15 ++++++++++++ sound/soc/soc-core.c | 68 ++++++++++++++++++++++++++++++++++++++++++++++------ 2 files changed, 76 insertions(+), 7 deletions(-) -- 1.8.5.2

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[alsa-devel] [RFC PATCH] ALSA: hda - Do not accept responses from non-existing codecs
by David Henningsson 29 Jan '14

29 Jan '14

While looking into some spurious responses, I found that the addr value was treated a bit inconsistent: values 8..0xf will be treated as codec 0 and values 0..7 will be treated as no error regardless of whether there is a codec there, or not. With this patch, all non-existing codecs will be treated equally. In addition, printing rp and wp could help figuring out if the wp value is reported wrongly from the controller or if something else is wrong. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 23 +++++++++-------------- 1 file changed, 9 insertions(+), 14 deletions(-) Note: this patch is untested (hence the RFC indication) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a..e87eac5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -834,18 +834,6 @@ static unsigned int azx_command_addr(u32 cmd) return addr; } -static unsigned int azx_response_addr(u32 res) -{ - unsigned int addr = res & 0xf; - - if (addr >= AZX_MAX_CODECS) { - snd_BUG(); - addr = 0; - } - - return addr; -} - /* send a command */ static int azx_corb_send_cmd(struct hda_bus *bus, u32 val) { @@ -907,8 +895,15 @@ static void azx_update_rirb(struct azx *chip) rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */ res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]); res = le32_to_cpu(chip->rirb.buf[rp]); - addr = azx_response_addr(res_ex); - if (res_ex & ICH6_RIRB_EX_UNSOL_EV) + addr = res_ex & 0xf; + if ((addr >= AZX_MAX_CODECS) || !(chip->codec_mask & (1 << addr))) { + snd_printk(KERN_ERR SFX "%s: spurious response %#x:%#x, rp = %d, wp = %d", + pci_name(chip->pci), + res, res_ex, + chip->rirb.rp, wp); + snd_BUG(); + } + else if (res_ex & ICH6_RIRB_EX_UNSOL_EV) snd_hda_queue_unsol_event(chip->bus, res, res_ex); else if (chip->rirb.cmds[addr]) { chip->rirb.res[addr] = res; -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Add parameter for dumping processing coefficients
by David Henningsson 29 Jan '14

29 Jan '14

Processing coefficients are often a vital part of the codec's configuration, so dumping them can be important. However, because they are undocumented and secret, we do not want to enable this for all codecs by default. Therefore instead add this as a debugging parameter. I have prepared for codecs that want to enable this by default by the extra dump_coef bitfield, but unsure if we want to do that as long as the (unlikely, but still) race remains. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_proc.c | 34 ++++++++++++++++++++++++++++++++-- 2 files changed, 33 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2b5d19e..ab2a444 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -361,6 +361,7 @@ struct hda_codec { unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ + unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index a8cb22e..ce5a6da 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -24,9 +24,14 @@ #include <linux/init.h> #include <linux/slab.h> #include <sound/core.h> +#include <linux/module.h> #include "hda_codec.h" #include "hda_local.h" +static int dump_coef = -1; +module_param(dump_coef, int, 0644); +MODULE_PARM_DESC(dump_coef, "Dump processing coefficients in codec proc file (-1=auto, 0=disable, 1=enable)"); + static char *bits_names(unsigned int bits, char *names[], int size) { int i, n; @@ -488,14 +493,39 @@ static void print_unsol_cap(struct snd_info_buffer *buffer, (unsol & AC_UNSOL_ENABLED) ? 1 : 0); } +static inline bool can_dump_coef(struct hda_codec *codec) +{ + switch (dump_coef) { + case 0: return false; + case 1: return true; + default: return codec->dump_coef; + } +} + static void print_proc_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { + unsigned int i, ncoeff, oldindex; unsigned int proc_caps = snd_hda_param_read(codec, nid, AC_PAR_PROC_CAP); + ncoeff = (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT; snd_iprintf(buffer, " Processing caps: benign=%d, ncoeff=%d\n", - proc_caps & AC_PCAP_BENIGN, - (proc_caps & AC_PCAP_NUM_COEF) >> AC_PCAP_NUM_COEF_SHIFT); + proc_caps & AC_PCAP_BENIGN, ncoeff); + + if (!can_dump_coef(codec)) + return; + + /* Note: This is racy - another process could run in parallel and change + the coef index too. */ + oldindex = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_COEF_INDEX, 0); + for (i = 0; i < ncoeff; i++) { + unsigned int val; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, i); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PROC_COEF, + 0); + snd_iprintf(buffer, " Coeff 0x%02x: 0x%04x\n", i, val); + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, oldindex); } static void print_conn_list(struct snd_info_buffer *buffer, -- 1.7.9.5

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