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

[alsa-devel] [PATCH] ALSA: firewire-lib: fix kerneldoc errors
by Takashi Sakamoto 18 Nov '14

18 Nov '14

Complete missing parameters, correct wrong reference, and add an explaination about the differences between the latest specification and our implementation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.h | 24 ++++++++++++++++++++++-- sound/firewire/cmp.c | 2 ++ 2 files changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 4823c08..e6e8926 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -23,7 +23,7 @@ * corresponds to the end of event in the packet. Out of IEC 61883. * @CIP_WRONG_DBS: Only for in-stream. The value of dbs is wrong in in-packets. * The value of data_block_quadlets is used instead of reported value. - * @SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is + * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first * packet is not continuous from an initial value. @@ -43,7 +43,27 @@ enum cip_flags { }; /** - * enum cip_sfc - a stream's sample rate + * enum cip_sfc - supported Sampling Frequency Codes (SFCs) + * @CIP_SFC_32000: 32,000 data blocks + * @CIP_SFC_44100: 44,100 data blocks + * @CIP_SFC_48000: 48,000 data blocks + * @CIP_SFC_88200: 88,200 data blocks + * @CIP_SFC_96000: 96,000 data blocks + * @CIP_SFC_176400: 176,400 data blocks + * @CIP_SFC_192000: 192,000 data blocks + * @CIP_SFC_COUNT: the number of supported SFCs + * + * These values are used to show nominal Sampling Frequency Code in + * Format Dependent Field (FDF) of AMDTP packet header. In IEC 61883-6:2002, + * this code means the number of events per second. Actually the code + * represents the number of data blocks transferred per second in an AMDTP + * stream. + * + * In IEC 61883-6:2005, some extensions were added to support more types of + * data such as 'One Bit LInear Audio', therefore the meaning of SFC became + * different depending on the types. + * + * Currently our implementation is compatible with IEC 61883-6:2002. */ enum cip_sfc { CIP_SFC_32000 = 0, diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index ba8df5a..ae3bc19 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -114,6 +114,7 @@ static int pcr_modify(struct cmp_connection *c, * cmp_connection_init - initializes a connection manager * @c: the connection manager to initialize * @unit: a unit of the target device + * @direction: input or output * @pcr_index: the index of the iPCR/oPCR on the target device */ int cmp_connection_init(struct cmp_connection *c, @@ -154,6 +155,7 @@ EXPORT_SYMBOL(cmp_connection_init); /** * cmp_connection_check_used - check connection is already esablished or not * @c: the connection manager to be checked + * @used: the pointer to store the result of checking the connection */ int cmp_connection_check_used(struct cmp_connection *c, bool *used) { -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs
by Charles Keepax 18 Nov '14

18 Nov '14

From: JS Park <aitdark.park(a)samsung.com> Signed-off-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 ++++++++++++++++++++---------- 1 files changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 08d8ec3..e54d7b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1053,8 +1053,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); @@ -1071,8 +1073,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); @@ -1101,8 +1105,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); @@ -1119,8 +1125,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); @@ -1137,8 +1145,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); -- 1.7.2.5

2 1

[alsa-devel] [PATCH 0/6] Add support for the WM8280 and WM8281 codecs
by Richard Fitzgerald 18 Nov '14

18 Nov '14

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. Richard Fitzgerald (6): mfd: arizona: add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 10 ++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 14 ++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 32 insertions(+), 8 deletions(-) -- 1.7.2.5

5 13

[alsa-devel] [PATCH] ASoC: rt286: build warning of section mismatch
by Sudip Mukherjee 18 Nov '14

18 Nov '14

From: Sudip Mukherjee <sudip(a)vectorindia.org> while building we were getting the following build warning: Section mismatch in reference from the function rt286_i2c_probe() to the variable .init.data:force_combo_jack_table The function rt286_i2c_probe() references the variable __initdata force_combo_jack_table. This is often because rt286_i2c_probe lacks a __initdata annotation or the annotation of force_combo_jack_table is wrong. we were getting the warning as force_combo_jack_table was marked with __initdata Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2e818aa..2cd4fe4 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); -static struct dmi_system_id force_combo_jack_table[] __initdata = { +static struct dmi_system_id force_combo_jack_table[] = { { .ident = "Intel Wilson Beach", .matches = { -- 1.8.1.2

2 1

[alsa-devel] [PATCH 00/11] ASoC: Untangle AC'97 support form the core
by Lars-Peter Clausen 18 Nov '14

18 Nov '14

The AC'97 support has had a special role within the ASoC framework with it being (in contrast to other audio buses) tightly integrated into the core. This patch series creates a clean separation between the core and the AC'97 support making AC'97 from the point of view of the core just another bus. Some of the changes in this series are required for the upcoming conversion of the AC'97 drivers to regmap, the other changes just fit nicely with the idea of componentization and having a generic ASoC core. The series starts with moving AC'97 support to its own file and doing a few general cleanups. After that it drops support of two features which were mainly responsible for the tight integration but are seemingly unused. This is a) Delaying the initialization and registration of the AC'97 device until after the card has been registered. There doesn't seem to be a good reason to delay to registration, all the information we need to register the device is available at the time when the device is created. Furthermore this will be needed for regmap. b) Setting platform data for the AC'97 via the CPU DAI. This hasn't been used in the last 5 years and doesn't seem to be that useful in general. If we should ever have a need for those features again they can still be implemented on top of the current code while keeping the clean separation. But for the time being it is simpler to just drop unused code rather than restructuring it. Finally the series pushes the pointer to the AC'97 device from the snd_soc_codec struct out to the driver private structs. Depends on asoc/topic/ad1980, asoc/topic/wm9705 and asoc/topic/wm971x - Lars Lars-Peter Clausen (11): ASoC: mpc5200_psc_ac97: Remove unused on-stack snd_ac97 device ASoC: mpc5200_dma: Don't overwrite ac97 device private_data ASoC: Properly handle AC'97 device lifetime management ASoC: Move AC'97 support to its own file ASoC: ac97: Use static ac97_bus ASoC: ac97: Merge soc_ac97_dev_{un,}register()/soc_{un,}register_ac97_codec() ASoC: ac97: Drop support for setting platform data via the CPU DAI ASoC: ac97: Drop delayed device registration ASoC: Drop ac97_control initialization from CODEC driver DAIs ASoC: Rename snd_soc_dai_driver struct ac97_control field to bus_control ASoC: ac97: Push snd_ac97 pointer to the driver level include/sound/soc-dai.h | 4 +- include/sound/soc.h | 25 ++- sound/soc/Makefile | 4 + sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/codecs/ac97.c | 18 +- sound/soc/codecs/ad1980.c | 28 +-- sound/soc/codecs/stac9766.c | 40 +++-- sound/soc/codecs/wm9705.c | 32 ++-- sound/soc/codecs/wm9712.c | 33 ++-- sound/soc/codecs/wm9713.c | 32 ++-- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 3 - sound/soc/fsl/mpc5200_psc_ac97.c | 6 +- sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/samsung/ac97.c | 4 +- sound/soc/sh/hac.c | 2 +- sound/soc/soc-ac97.c | 256 +++++++++++++++++++++++++++ sound/soc/soc-core.c | 372 ++------------------------------------- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- 25 files changed, 428 insertions(+), 455 deletions(-) create mode 100644 sound/soc/soc-ac97.c -- 1.8.0

3 13

[alsa-devel] [PATCH 1/2] ALSA: hda - move DELL_WMI_MIC_MUTE_LED to the tail in the quirk chain
by Hui Wang 18 Nov '14

18 Nov '14

We have one more Dell machine needs DELL_WMI_MIC_MUTE_LED quirk, but the machine uses alc293 instead of alc255. So if DELL_WMI_MIC_MUTE_LED still chain ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, the machine can't use this quirk. To change this situation, let the DELL_WMI_MIC_MUTE_LED to be a standalone quirk, and let other quirks chain it. After this change, this quirk can be chained to any existing quirks, and as a result, it is possible that this quirk is applied to a non-Dell machine or a Dell machine without mic mute led on it, but it is still safe since alc_fixup_dell_wmi() will return an error in these situations. And remove the quirk for machine with subsystem id 0x6010 and 0x601f, these two machines will fall back to the quirk ALC255_FIXUP_DELL1_MIC_NO_PRESENCE-->ALC255_FIXUP_HEADSET_MODE--> ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED through pin_fixup_tbl[]. BugLink: https://bugs.launchpad.net/bugs/1381856 Reported-and-Tested-by: Po-Hsu Lin <po-hsu.lin(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1723954..9da5798 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4709,6 +4709,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, + .chained = true, + .chain_id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED }, [ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC] = { .type = HDA_FIXUP_FUNC, @@ -4744,8 +4746,6 @@ static const struct hda_fixup alc269_fixups[] = { [ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_wmi, - .chained_before = true, - .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, [ALC282_FIXUP_ASPIRE_V5_PINS] = { .type = HDA_FIXUP_PINS, @@ -4783,10 +4783,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), - SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: hda: separate stream tag for playback and capture
by Vinod Koul 18 Nov '14

18 Nov '14

From: Rafal Redzimski <rafal.f.redzimski(a)intel.com> According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. Newer Intel HW has the capability to provides a total of more than 15 stream DMA engines which with the current implementation causes an overflow on STRM field (and the whole register) which results in usage of reserved value 0 in the STRM field which confuses HDA controller. So make stream_tag assignment separate for input and output streams Signed-off-by: Rafal Redzimski <rafal.f.redzimski(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..703ac6e 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1921,11 +1921,17 @@ int azx_mixer_create(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return index < chip->playback_index_offset; +} /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1944,13 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout the stream direction + * group, valid values 1...15 + */ + azx_dev->stream_tag = is_input_stream(chip, i) ? + ++in_stream_tag : ++out_stream_tag; } return 0; -- 1.7.0.4

2 1

[alsa-devel] ASoC: omap-mcbsp: Deletion of an unnecessary check before the function call "kfree"
by SF Markus Elfring 18 Nov '14

18 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 14:05:27 +0100 The kfree() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- 2.1.3

3 2

[alsa-devel] [PATCH] ASoC: fsl_ssi: Remove comment about SSI running only in slave mode
by Fabio Estevam 18 Nov '14

18 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Current driver can also run in I2S master mode, so remove the old comment. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ab05262..1ef0c84 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,8 +67,6 @@ /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI * - * This driver currently only supports the SSI running in I2S slave mode. - * * The SSI has a limitation in that the samples must be in the same byte * order as the host CPU. This is because when multiple bytes are written * to the STX register, the bytes and bits must be written in the same -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rt5677: Modify the default value of the MX-8E[4] for ASRC function
by Oder Chiou 18 Nov '14

18 Nov '14

Modify the default value of the MX-8E[4] to 1 for ASRC function. It could prevent the pop noise with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d659237..49ac9ba 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,7 +55,8 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, {RT5677_PR_BASE + 0x13, 0x0312}, {RT5677_PR_BASE + 0x1e, 0x0000}, @@ -173,7 +174,7 @@ static const struct reg_default rt5677_reg[] = { {RT5677_ASRC_9 , 0x0000}, {RT5677_ASRC_10 , 0x0000}, {RT5677_ASRC_11 , 0x0000}, - {RT5677_ASRC_12 , 0x0008}, + {RT5677_ASRC_12 , 0x0018}, {RT5677_ASRC_13 , 0x0000}, {RT5677_ASRC_14 , 0x0000}, {RT5677_ASRC_15 , 0x0000}, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] ASoC fixes for v3.18
by Mark Brown 17 Nov '14

17 Nov '14

The following changes since commit fc14f9c1272f62c3e8d01300f52467c0d9af50f9: Linux 3.18-rc5 (2014-11-16 16:36:20 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18-rc5 for you to fetch changes up to cf9a7f7823c67243da44da2ac47ca944a3108282: Merge remote-tracking branches 'asoc/fix/rt5670', 'asoc/fix/samsung' and 'asoc/fix/sgtl5000' into asoc-linus (2014-11-17 16:41:11 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.18 As well as the usual driver fixes there's a few other things here: One is a fix for a race in DPCM which is unfortuantely a rather large diffstat, this is the result of growing usage of the mainline code and hence more detailed testing so I'm relatively happy. The other is a fix for non-DT machine driver matching following some of the componentization work which is much more focused. Both have had a while to cook in -next. ---------------------------------------------------------------- Andreas Färber (1): ASoC: samsung: Add MODULE_DEVICE_TABLE for Snow Axel Lin (1): ASoC: es8328-i2c: Fix i2c_device_id name field in es8328_id Bard Liao (2): ASoC: rt5670: correct the incorrect default values ASoC: rt5670: change dapm routes of PLL connection Charles Keepax (1): ASoC: wm_adsp: Avoid attempt to free buffers that might still be in use Dylan Reid (1): ASoC: max98090: Correct pclk divisor settings Fabio Estevam (1): ASoC: sgtl5000: Fix SMALL_POP bit definition Fang, Yang A (1): ASoC: rt5645: Mark RT5645_TDM_CTRL_3 as readable Jianqun (1): ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_rxctrl Kuninori Morimoto (2): ASoC: fsi: remove unsupported PAUSE flag ASoC: rsnd: remove unsupported PAUSE flag Lars-Peter Clausen (1): ASoC: Fix snd_soc_find_dai() matching component by name Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/cs41l51', 'asoc/fix/dpcm', 'asoc/fix/es8328', 'asoc/fix/fsl-asrc', 'asoc/fix/max98090', 'asoc/fix/rcar', 'asoc/fix/rockchip' and 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5670', 'asoc/fix/samsung' and 'asoc/fix/sgtl5000' into asoc-linus Nicolin Chen (1): ASoC: fsl_asrc: Add reg_defaults for regmap to fix kernel dump Takashi Iwai (1): ASoC: dpcm: Fix race between FE/BE updates and trigger Thomas Petazzoni (1): ASoC: cs42l51: re-hook of_match_table pointer include/sound/soc-dpcm.h | 2 ++ sound/soc/codecs/cs42l51-i2c.c | 1 + sound/soc/codecs/cs42l51.c | 4 ++- sound/soc/codecs/cs42l51.h | 1 + sound/soc/codecs/es8328-i2c.c | 2 +- sound/soc/codecs/max98090.c | 6 ++-- sound/soc/codecs/rt5645.c | 2 ++ sound/soc/codecs/rt5670.c | 36 ++++++++++---------- sound/soc/codecs/sgtl5000.c | 3 +- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/wm_adsp.c | 1 + sound/soc/fsl/fsl_asrc.c | 26 ++++++++++++++ sound/soc/rockchip/rockchip_i2s.c | 4 ++- sound/soc/samsung/snow.c | 1 + sound/soc/sh/fsi.c | 3 +- sound/soc/sh/rcar/core.c | 3 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 72 ++++++++++++++++++++++++++++++--------- 18 files changed, 123 insertions(+), 48 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: samsung-i2s: Add 192kHz config option
by Richard Fitzgerald 17 Nov '14

17 Nov '14

Adds QUIRK_SUPPORTS_192KHZ to allow 192kHz rate to be selected for hardware that supports it. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 2 ++ include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s.c | 8 ++++++++ 3 files changed, 11 insertions(+), 0 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index d188296..c3f3950 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -40,6 +40,8 @@ Optional SoC Specific Properties: sub system(used in secondary sound source). - pinctrl-0: Should specify pin control groups used for this controller. - pinctrl-names: Should contain only one value - "default". +- samsung,supports-192khz: specify this (without a value) if you want to allow + 192000 sample rate over the I2S link. Example: diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 5e0bc77..6d0702d 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -28,6 +28,7 @@ struct samsung_i2s { #define QUIRK_NEED_RSTCLR (1 << 3) #define QUIRK_SUPPORTS_TDM (1 << 4) #define QUIRK_SUPPORTS_IDMA (1 << 5) +#define QUIRK_SUPPORTS_192KHZ (1 << 6) /* Quirks of the I2S controller */ u32 quirks; dma_addr_t idma_addr; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 947352d..f688e89 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1219,6 +1219,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) "specified"); } } + + if (of_find_property(np, "samsung,supports-192khz", NULL)) + quirks |= QUIRK_SUPPORTS_192KHZ; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1247,6 +1250,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (quirks & QUIRK_PRI_6CHAN) pri_dai->i2s_dai_drv.playback.channels_max = 6; + if (quirks & QUIRK_SUPPORTS_192KHZ) { + pri_dai->i2s_dai_drv.playback.rates |= SNDRV_PCM_RATE_192000; + pri_dai->i2s_dai_drv.capture.rates |= SNDRV_PCM_RATE_192000; + } + if (quirks & QUIRK_SEC_DAI) { sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Avoid attempt to free buffers that might still be in use
by Charles Keepax 17 Nov '14

17 Nov '14

We should not free any buffers associated with writing out coefficients to the DSP until all the async writes have completed. This patch updates the out of memory path when allocating a new buffer to include a call to regmap_async_complete. Reported-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0a08ef5e..08d8ec3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1343,6 +1343,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) file, blocks, pos - firmware->size); out_fw: + regmap_async_complete(regmap); release_firmware(firmware); wm_adsp_buf_free(&buf_list); out: -- 1.7.2.5

2 1

[alsa-devel] [PATCH] usb-audio: Set the Control Selector to SU_SELECTOR_CONTROL for UAC2
by Johan Rastén 17 Nov '14

17 Nov '14

Specified in section 5.2.5.6.1 of the USB Audio Class 2.0 definition. Solves the following error for C-Media 6632A (Asus Xonar U7): [ 8219.676164] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 Signed-off-by: Johan Rastén <johan(a)oljud.se> --- sound/usb/mixer.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 14e1455..c6209ce 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2032,10 +2032,11 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, cval->res = 1; cval->initialized = 1; - if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) - cval->control = UAC2_CX_CLOCK_SELECTOR; - else + if (state->mixer->protocol == UAC_VERSION_1) cval->control = 0; + else /* UAC_VERSION_2 */ + cval->control = (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) ? + UAC2_CX_CLOCK_SELECTOR : UAC2_SU_SELECTOR; namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); if (!namelist) { -- 2.1.0

2 1

[alsa-devel] [PATCH (alsa-lib)] USB-audio: Add five more cards to IEC958 blacklist
by David Henningsson 17 Nov '14

17 Nov '14

These devices do not have any IEC958 outputs, so prevent them from being opened. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index a6754d9..05e5d09 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -38,10 +38,14 @@ USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 # The below don't have digital in/out, so prevent them from being opened. + "Andrea PureAudio USB-SA Headset" 999 "Blue Snowball" 999 "HP Digital Stereo Headset" 999 + "GN 9330" 999 "Logitech Speaker Lapdesk N700" 999 + "Logitech G35 Headset" 999 "Logitech USB Headset" 999 + "Logitech USB Headset H540" 999 "Logitech Wireless Headset" 999 "Plantronics GameCom 780" 999 "Plantronics USB Headset" 999 @@ -50,6 +54,7 @@ USB-Audio.pcm.iec958_device { "Scarlett 2i2 USB" 999 "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 + "SWTOR Gaming Headset by Razer" 999 "USB Device 0x46d:0x992" 999 } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] ALSA: powermac: Deletion of an unnecessary check before the function call "pci_dev_put"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 13:35:54 +0100 The pci_dev_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/ppc/pmac.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 8a431bc..5a13b22 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -887,8 +887,7 @@ static int snd_pmac_free(struct snd_pmac *chip) } } - if (chip->pdev) - pci_dev_put(chip->pdev); + pci_dev_put(chip->pdev); of_node_put(chip->node); kfree(chip); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: hdsp: Deletion of an unnecessary check before the function call "release_firmware"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 13:04:14 +0100 The release_firmware() function tests whether its argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/rme9652/hdsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7646ba1..0ae568d 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5368,8 +5368,7 @@ static int snd_hdsp_free(struct hdsp *hdsp) snd_hdsp_free_buffers(hdsp); - if (hdsp->firmware) - release_firmware(hdsp->firmware); + release_firmware(hdsp->firmware); vfree(hdsp->fw_uploaded); if (hdsp->iobase) -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: lola: Deletion of an unnecessary check before the function call "vfree"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 12:42:16 +0100 The vfree() function performs also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/lola/lola_mixer.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index 782f4d8..e7fe15d 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -108,8 +108,7 @@ int lola_init_pins(struct lola *chip, int dir, int *nidp) void lola_free_mixer(struct lola *chip) { - if (chip->mixer.array_saved) - vfree(chip->mixer.array_saved); + vfree(chip->mixer.array_saved); } int lola_init_mixer_widget(struct lola *chip, int nid) -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: ice17xx: Deletion of unnecessary checks before the function call "snd_ac97_resume"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 11:28:02 +0100 The snd_ac97_resume() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ice1712/ice1712.c | 3 +-- sound/pci/ice1712/ice1724.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 87f7fc4..e1e18b5 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2905,8 +2905,7 @@ static int snd_ice1712_resume(struct device *dev) outw(ice->pm_saved_spdif_ctrl, ICEMT(ice, ROUTE_SPDOUT)); outw(ice->pm_saved_route, ICEMT(ice, ROUTE_PSDOUT03)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 08cb08a..0e56835 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2884,8 +2884,7 @@ static int snd_vt1724_resume(struct device *dev) outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG)); outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda: Deletion of unnecessary checks before two function calls
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 10:44:33 +0100 The functions kfree() and release_firmware() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/hda_intel.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec6a7d0..3fe8859 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -827,8 +827,7 @@ static void snd_hda_bus_free(struct hda_bus *bus) WARN_ON(!list_empty(&bus->codec_list)); if (bus->workq) flush_workqueue(bus->workq); - if (bus->unsol) - kfree(bus->unsol); + kfree(bus->unsol); if (bus->ops.private_free) bus->ops.private_free(bus); if (bus->workq) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a7f848..b4ec4e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1129,8 +1129,7 @@ static int azx_free(struct azx *chip) pci_disable_device(chip->pci); kfree(chip->azx_dev); #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (chip->fw) - release_firmware(chip->fw); + release_firmware(chip->fw); #endif if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(false); -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: pxa: prepare/unprepare clocks in pxa-ssp
by Dmitry Eremin-Solenikov 17 Nov '14

17 Nov '14

Change clk_enable/disable() calls to clk_prepare_enable() and clk_disable_unrepapre(). Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a8e0974..cbba063 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -97,7 +97,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); pxa_ssp_disable(ssp); } @@ -121,7 +121,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); } kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); @@ -136,7 +136,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); @@ -144,7 +144,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->psp = __raw_readl(ssp->mmio_base + SSPSP); pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -154,7 +154,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); __raw_writel(sssr, ssp->mmio_base + SSSR); __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); @@ -165,7 +165,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) if (cpu_dai->active) pxa_ssp_enable(ssp); else - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -256,11 +256,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (ssp->type != PXA3xx_SSP) - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); if (ssp->type != PXA3xx_SSP) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); return 0; } -- 2.1.1

3 2

[alsa-devel] [PATCH] ASoC: Intel: add missing ACPI device table
by mengdong.lin＠intel.com 17 Nov '14

17 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The ACPI device table will generate the driver module alias for Intel audio devices enumerated from ACPI. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 2b1c5d9..b261821 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -342,6 +342,8 @@ static const struct acpi_device_id sst_acpi_ids[] = { { }, }; +MODULE_DEVICE_TABLE(acpi, sst_acpi_ids); + static struct platform_driver sst_acpi_driver = { .driver = { .name = "intel_sst_acpi", -- 1.9.1

3 2

[alsa-devel] [PATCH 1/3] ASoC: adav80x: Replace w->codec with snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 17 Nov '14

17 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adav80x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index ce3cdca..b67480f 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -212,7 +212,7 @@ static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); const char *clk; @@ -236,7 +236,7 @@ static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; -- 1.8.0

2 5

[alsa-devel] TLV320AIC3106 + DM3730 soc doens't capture audio (no BLCK, no WLCK)
by Sergio Serrano 17 Nov '14

17 Nov '14

Dear members, Please, let me explain my setup: I'm using DM3730 soc <https://www.isee.biz/products/igep-processor-boards/igepv2-dm3730> with McBSP3 and I2C2 to drive audio and control to the TLV320AIC3106 codec (which it is acting as master). Currently I'm using 2.6.37 kernel version <http://git.isee.biz/?p=pub/scm/linux-omap-2.6.git;a=tree>. For reference I've used BBB Audio Cape <http://elinux.org/CircuitCo:Audio_Cape_RevA> as base design. Right now I've got playback working as expected (codec generates BCLK and WCLK). Whenever I try to record using "arecor,d -f cd -Dhw:1,0" it's only showing wave header but no audio is captured at all. I can not see any clocks running on the codec. How can I debug or go further with this issue? Any thoughts will be appreciated!! BTW this is the init code: static int TLV320AIC3106_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; printk(KERN_ERR "+tlv_hw_params(%p,%p) %p, %p, %p\n", substream, params, rtd, codec_dai, cpu_dai); // Set codec DAI configuration ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set codec DAI configuration\n"); return ret; } // Set cpu DAI configuration ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set cpu DAI configuration\n"); return ret; } // Set the codec system clock for DAC and ADC ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set codec system clock\n"); return ret; } printk(KERN_ERR "-tlv_hw_params\n"); return 0; } BR, Sergio

1 0

[alsa-devel] Pop and offsets at start of audio playback/record for SGTL5000 on i.MX28
by Craig McQueen 17 Nov '14

17 Nov '14

I'm testing the 3.14.19 kernel for i.MX28 EVK, which has an SGTL5000 CODEC. I've also tested on the 3.18-rc4 kernel and confirmed this issue still occurs. Playback When doing audio playback, I notice that the audio "fades in" over the first (approximately) 500 ms of playback. This is very noticeable and not ideal in certain applications (e.g. audio notifications that are of short duration). However, if audio playback starts while audio recording is already in-progress, then there is no fade-in; instead there is a loud pop. I assume the fade-in is there to mask the pop. The pop is very unpleasant through headphones. (However, if I do audio playback a _second_ time while audio recording is still in-progress, then there is no loud pop the second time.) I didn't think either the pop or the fade-in occurred on the 2.6.35 kernel. How feasible is it to improve the driver to avoid or reduce the pop, and remove the necessity for a fade-in? Record When recording, I notice there is a low-frequency ramping offset at the start of a recording (after the 400-500 ms of previous audio) of approx 600-700 ms duration. This also reduces the audio quality at the start of the recording. It could reduce the quality of captured audio depending on the application usage. I am getting distortion (clipping I assume) of my audio input during this time. It could cause a pop on playback (local playback or e.g. VoIP transmission to a remote device). It could reduce the audio quality through an audio CODEC. How feasible is it to improve the driver to avoid a ramping offset at the start of recording? -- Craig McQueen

3 6

[alsa-devel] What tree to base patches on?
by Johan Aires Rastén 16 Nov '14

16 Nov '14

If I want to send you a patch for snd_usb_audio, is there a git repo + branch or something else which I should base it on?

2 1

[alsa-devel] [PATCH v2] ASoC: Fix kerneldoc errors
by Takashi Iwai 16 Nov '14

16 Nov '14

Lots of typos and missing parameter descriptions. Some private struct fields are moved to the end of struct and put markers to make kerneldoc happy (and it make even easier for human readers). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 16 +++++++++++----- sound/soc/soc-core.c | 21 ++++++++++++++------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-devres.c | 2 +- sound/soc/soc-jack.c | 2 +- 5 files changed, 28 insertions(+), 15 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130037a0..10129be3cbb1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -588,10 +588,11 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @invert: if non-zero then pin is enabled when status is not reported */ struct snd_soc_jack_pin { - struct list_head list; const char *pin; int mask; bool invert; + /* private: */ + struct list_head list; }; /** @@ -602,13 +603,13 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container */ struct snd_soc_jack_zone { unsigned int min_mv; unsigned int max_mv; unsigned int jack_type; unsigned int debounce_time; + /* private: */ struct list_head list; }; @@ -618,16 +619,19 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debouce time in ms * @wake: enable as wake source + * @jack: assigned jack + * @desc: GPIO descriptor * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an * ADC). + * @data: arbitrary data passed to jack_status_check callback */ struct snd_soc_jack_gpio { unsigned int gpio; @@ -640,11 +644,13 @@ struct snd_soc_jack_gpio { bool wake; struct snd_soc_jack *jack; - struct delayed_work work; struct gpio_desc *desc; void *data; int (*jack_status_check)(void *data); + + /* private: */ + struct delayed_work work; }; struct snd_soc_jack { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a23a0e9..74a8a80e2d9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2113,6 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: AC97 opts to set + * @pdev: platform to deal with * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. @@ -2685,7 +2687,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -3458,7 +3460,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -4082,6 +4084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core + * @dev: device to unregister * */ void snd_soc_unregister_component(struct device *dev) @@ -4119,7 +4122,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -4158,7 +4161,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: platform to register + * @platform_drv: the driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -4215,7 +4219,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -4306,7 +4310,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: codec to register + * @codec_drv: the codec driver to register + * @dai_drv: the array of DAI drivers to register + * @num_dai: the number of DAIs to register */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -4406,7 +4413,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..7050ec1d5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2680,7 +2680,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card object containing widgets * * Checks the codec for any new dapm widgets and creates them if found. * diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..ef1d42d7c6f6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- 2.1.3

2 7

[alsa-devel] [PATCH] usb-audio: Add ctrl message delay quirk for Marantz/Denon devices
by Jurgen Kramer 16 Nov '14

16 Nov '14

This patch adds a USB control message delay quirk for a few specific Marantz/Denon devices. Without the delay the DACs will not work properly and produces the following type of messages: Nov 15 10:09:21 orwell kernel: [ 91.342880] usb 3-13: clock source 41 is not valid, cannot use Nov 15 10:09:21 orwell kernel: [ 91.343775] usb 3-13: clock source 41 is not valid, cannot use There are likely other Marantz/Denon devices using the same USB module which exhibit the same problems. But as this cannot be verified I limited the patch to the devices I could test. The following two devices are covered by this path: - Marantz SA-14S1 - Marantz HD-DAC1 Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d2aa45a..a5941f8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1146,6 +1146,20 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); + + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + + switch (le16_to_cpu(dev->descriptor.idProduct)) { + case 0x3005: /* Marantz HD-DAC1 */ + case 0x3006: /* Marantz SA-14S1 */ + mdelay(20); + break; + } + } } /* -- 1.9.3

2 1

[alsa-devel] [PATCH] snd_dlsym: only do versioning checks if built --with-versioning
by John Spencer 16 Nov '14

16 Nov '14

From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/dlmisc.c b/src/dlmisc.c index 4b8a02c..62f7047 100644 --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -177,11 +177,13 @@ void *snd_dlsym(void *handle, const char *name, const char *version) } #endif #ifdef HAVE_LIBDL +#ifdef VERSIONED_SYMBOLS if (version) { err = snd_dlsym_verify(handle, name, version); if (err < 0) return NULL; } +#endif return dlsym(handle, name); #else return NULL; -- 1.8.4

2 1

[alsa-devel] DTS-HD over HDMI
by Kip Warner 15 Nov '14

15 Nov '14

Hey list, I apologize if this ought to have gone to alsa-user. I wasn't sure as a developer using API, but not developing ALSA itself, if it have been better there than here. I'd like to be able to stream DTS-HD Master Audio bits out the HDMI port in their compressed form, and not decompressed as LPCM. From what I can gather, this may be possible with the ALSA API, provided the HDMI port is >= 1.3 and the audio card supports high bit-rate (HBR). Is my thinking correct? -- Kip Warner -- Senior Software Engineer OpenPGP encrypted/signed mail preferred http://www.thevertigo.com

2 2

[alsa-devel] usb-audio: check_input_term (mixer.c) iterating to clock selectors?
by Johan Aires Rastén 15 Nov '14

15 Nov '14

Hi everybody! I've just started looking at ALSA because I want to get the mic working with my Asus Xonar U7. Some of you might have seen me on IRC as johanar. I'm wondering about check_input_term(...) in sound/usb/mixer.c, which for case UAC_INPUT_TERMINAL and UAC_VERSION_2 will recursively go into the clock source. Is this correct behaviour? For my card both mic and line will show up as "Unit 22" when I call "amixer contents" since the two inputs have the same unnamed clock source (they also share physical connector). However, if I comment out the recursive call they show up as "mic" and "line" as they're supposed to. Btw, if this is correct, couldn't we set id = d->bCSourceID and break; in the same way that case UAC_FEATURE_UNIT does? //Johan Rastén

1 0

[alsa-devel] [PATCH 1/2] spi: Fix mapping from vmalloc-ed buffer to scatter list
by Charles Keepax 14 Nov '14

14 Nov '14

We can only use page_address on memory that has been mapped using kmap, when the buffer passed to the SPI has been allocated by vmalloc the page has not necessarily been mapped through kmap. This means sometimes page_address will return NULL causing the pointer we pass to sg_set_buf to be invalid. As we only call page_address such that we can pass a virtual address to sg_set_buf which will then immediately call virt_to_page on it, fix this by calling sg_set_page directly rather then relying on the sg_set_buf helper. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/spi/spi.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/spi/spi.c b/drivers/spi/spi.c index ebcb33d..50f20f2 100644 --- a/drivers/spi/spi.c +++ b/drivers/spi/spi.c @@ -615,13 +615,13 @@ static int spi_map_buf(struct spi_master *master, struct device *dev, sg_free_table(sgt); return -ENOMEM; } - sg_buf = page_address(vm_page) + - ((size_t)buf & ~PAGE_MASK); + sg_set_page(&sgt->sgl[i], vm_page, + min, offset_in_page(buf)); } else { sg_buf = buf; + sg_set_buf(&sgt->sgl[i], sg_buf, min); } - sg_set_buf(&sgt->sgl[i], sg_buf, min); buf += min; len -= min; -- 1.7.2.5

2 7

[alsa-devel] Question on Compressed offload session
by gsantosh＠codeaurora.org 14 Nov '14

14 Nov '14

Hi All, The Question is for the compressed offload session. For a generic codec driver during the startup function it will set some of the hw_constraints rule similarly like this. snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_12_24); pcm_lib.c will try to add the rule to the runtime structure by accessing the pointers which will be initialized during opening of the session, as The Constraints added by the codec driver will be updated in the struct snd_pcm_hw_constraints of runtime structure which will be part of substream handle. But for the compressed offload I do not see the initialization done for HW constraints, as done in pcm session 2092int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, 2093 struct file *file, 2094 struct snd_pcm_substream **rsubstream) most of the existing drivers which has the hw_constraint_list code will not be applicable for compress offload session, how to solve this? Regards, Santosh M G. Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,a Linux Foundation Collaborative Project

3 4

[alsa-devel] (no subject)
by Angelo Dureghello 14 Nov '14

14 Nov '14

unsubscribe alsa-devel angelo70(a)gmail.com -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH 00/15] new locomo driver
by Dmitry Eremin-Solenikov 14 Nov '14

14 Nov '14

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. Current drivers present in Linux kernel has some problems: * It uses custom bus instead of platform bus/mfd core. * Device drivers are not well layered/separated. * It uses custom gpio accessors instead of using GPIOLIB. With this patchset I tried to modernise and restructure the LoCoMo driver. Also I added an experimental SPI driver that will be used by mmc-spi on Sharp SL-5500 collie devices. ---------------------------------------------------------------- Dmitry Eremin-Solenikov (15): mfd: add new driver for Sharp LoCoMo GPIO: port LoCoMo gpio support from old driver leds: port locomo leds driver to new locomo core input: convert LoCoMo keyboard driver to use new locomo core video: backlight: add new locomo backlight driver video: lcd: add LoCoMo LCD driver video: backlight: drop old locomo bl/lcd driver ARM: sa1100: make collie use new locomo drivers ARM: sa1100: don't preallocate IRQ space for locomo ARM: pxa: poodle: use new LoCoMo driver sound: soc: poodle: make use of new locomo GPIO interface ARM: pxa: poodle: don't preallocate IRQ space for locomo ARM: drop old LoCoMo driver gpio: locomo: implement per-pin irq handling spi: add locomo SPI driver arch/arm/common/Kconfig | 3 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 914 --------------------- arch/arm/mach-pxa/Kconfig | 1 - arch/arm/mach-pxa/include/mach/poodle.h | 16 +- arch/arm/mach-pxa/poodle.c | 23 +- arch/arm/mach-sa1100/Kconfig | 1 - arch/arm/mach-sa1100/collie.c | 112 +-- arch/arm/mach-sa1100/include/mach/collie.h | 25 +- arch/arm/mach-sa1100/include/mach/irqs.h | 19 +- drivers/gpio/Kconfig | 7 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-locomo.c | 386 +++++++++ drivers/input/keyboard/Kconfig | 1 - drivers/input/keyboard/locomokbd.c | 165 ++-- drivers/leds/Kconfig | 1 - drivers/leds/leds-locomo.c | 104 ++- drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/locomo.c | 644 +++++++++++++++ drivers/spi/Kconfig | 8 + drivers/spi/Makefile | 1 + drivers/spi/spi-locomo.c | 370 +++++++++ drivers/video/backlight/Kconfig | 14 +- drivers/video/backlight/Makefile | 3 +- drivers/video/backlight/locomo_bl.c | 171 ++++ drivers/video/backlight/locomo_lcd.c | 224 +++++ drivers/video/backlight/locomolcd.c | 255 ------ .../asm/hardware => include/linux/mfd}/locomo.h | 148 ++-- sound/soc/pxa/poodle.c | 51 +- 30 files changed, 2185 insertions(+), 1493 deletions(-) delete mode 100644 arch/arm/common/locomo.c create mode 100644 drivers/gpio/gpio-locomo.c create mode 100644 drivers/mfd/locomo.c create mode 100644 drivers/spi/spi-locomo.c create mode 100644 drivers/video/backlight/locomo_bl.c create mode 100644 drivers/video/backlight/locomo_lcd.c delete mode 100644 drivers/video/backlight/locomolcd.c rename {arch/arm/include/asm/hardware => include/linux/mfd}/locomo.h (55%)

9 53

[alsa-devel] [PATCH 1/4] ASoC: rt5670: make bias level more reasonable
by bardliao＠realtek.com 14 Nov '14

14 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c4c42d5..85dc210 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2309,6 +2309,8 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2330,16 +2332,27 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); - snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); - snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); - snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); - snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); snd_soc_update_bits(codec, RT5670_PWR_ANLG1, RT5670_LDO_SEL_MASK, 0x1); break; + case SND_SOC_BIAS_OFF: + if (rt5670->pdata.jd_mode) + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_MB | RT5670_PWR_BG); + else + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + break; default: break; -- 1.8.1.1.439.g50a6b54

4 7

[alsa-devel] [PATCH v4] ALSA: ice1712: consider error value
by Sudip Mukherjee 14 Nov '14

14 Nov '14

earlier we were ignoring the return value of snd_ak4114_create and always returning 0. now we are returning the actual status. revo_init is calling this function, and revo_init is checking the return value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v3: spec->ak4114 is now dereferenced if we dont get an error value from snd_ak4114_create. change in v4: error value returned according to coding style. sound/pci/ice1712/revo.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1112ec1..1d81ae6 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -494,11 +494,13 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice) ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &spec->ak4114); + if (err < 0) + return err; /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; - return 0; /* error ignored; it's no fatal error */ + return 0; } static int revo_init(struct snd_ice1712 *ice) -- 1.8.1.2

2 4

[alsa-devel] [PATCH v3] ALSA: ice1712: consider error value
by Sudip Mukherjee 14 Nov '14

14 Nov '14

earlier we were ignoring the return value of snd_ak4114_create and always returning 0. now we are returning the actual status. revo_init is calling this function, and revo_init is checking the return value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v3: spec->ak4114 is now dereferenced if we dont get an error value from snd_ak4114_create. sound/pci/ice1712/revo.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1112ec1..24a7c6a 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -496,9 +496,10 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice) ice, &spec->ak4114); /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ - spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; + if (err == 0) + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; - return 0; /* error ignored; it's no fatal error */ + return err; } static int revo_init(struct snd_ice1712 *ice) -- 1.8.1.2

2 1

[alsa-devel] [PATCH v4] ASoC: Intel: mrfld- add ACPI module
by Vinod Koul 14 Nov '14

14 Nov '14

Add the last ACPI module support which also uses core module like the PCI part Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 5 + sound/soc/intel/sst/Makefile | 4 +- sound/soc/intel/sst/sst.c | 22 ++- sound/soc/intel/sst/sst.h | 1 + sound/soc/intel/sst/sst_acpi.c | 362 ++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 6 files changed, 391 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/sst/sst_acpi.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c963a5d..a992e85 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -20,6 +20,11 @@ config SND_SST_IPC_PCI tristate select SND_SST_IPC +config SND_SST_IPC_ACPI + tristate + select SND_SST_IPC + depends on ACPI + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI diff --git a/sound/soc/intel/sst/Makefile b/sound/soc/intel/sst/Makefile index b8aa1d3..fd21726 100644 --- a/sound/soc/intel/sst/Makefile +++ b/sound/soc/intel/sst/Makefile @@ -1,7 +1,7 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o snd-intel-sst-pci-objs += sst_pci.o - +snd-intel-sst-acpi-objs += sst_acpi.o obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o - +obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index b97c231..b2b5604 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -26,6 +26,7 @@ #include <linux/pm_runtime.h> #include <linux/pm_qos.h> #include <linux/async.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/soc.h> #include <asm/platform_sst_audio.h> @@ -181,6 +182,7 @@ int sst_driver_ops(struct intel_sst_drv *sst) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst->tstamp = SST_TIME_STAMP_MRFLD; sst->ops = &mrfld_ops; return 0; @@ -323,7 +325,7 @@ EXPORT_SYMBOL_GPL(sst_context_init); void sst_context_cleanup(struct intel_sst_drv *ctx) { pm_runtime_get_noresume(ctx->dev); - pm_runtime_forbid(ctx->dev); + pm_runtime_disable(ctx->dev); sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); @@ -371,8 +373,19 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); pm_runtime_use_autosuspend(ctx->dev); - pm_runtime_allow(ctx->dev); - pm_runtime_put_noidle(ctx->dev); + /* + * For acpi devices, the actual physical device state is + * initially active. So change the state to active before + * enabling the pm + */ + if (acpi_disabled) { + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); + } else { + pm_runtime_allow(ctx->dev); + pm_runtime_put_noidle(ctx->dev); + } + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); @@ -395,6 +408,9 @@ static int intel_sst_runtime_suspend(struct device *dev) synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); + /* save the shim registers because PMC doesn't save state */ + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); + return ret; } diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 2dcbf47..683dc71 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -29,6 +29,7 @@ /* driver names */ #define SST_DRV_NAME "intel_sst_driver" #define SST_MRFLD_PCI_ID 0x119A +#define SST_BYT_ACPI_ID 0x80860F28 #define SST_SUSPEND_DELAY 2000 #define FW_CONTEXT_MEM (64*1024) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c new file mode 100644 index 0000000..2b1c5d9 --- /dev/null +++ b/sound/soc/intel/sst/sst_acpi.c @@ -0,0 +1,362 @@ +/* + * sst_acpi.c - SST (LPE) driver init file for ACPI enumeration. + * + * Copyright (c) 2013, Intel Corporation. + * + * Authors: Ramesh Babu K V <Ramesh.Babu(a)intel.com> + * Authors: Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * + */ + +#include <linux/module.h> +#include <linux/fs.h> +#include <linux/interrupt.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/miscdevice.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/pm_runtime.h> +#include <linux/pm_qos.h> +#include <linux/acpi.h> +#include <asm/platform_sst_audio.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/compress_driver.h> +#include <acpi/acbuffer.h> +#include <acpi/platform/acenv.h> +#include <acpi/platform/aclinux.h> +#include <acpi/actypes.h> +#include <acpi/acpi_bus.h> +#include "../sst-mfld-platform.h" +#include "../sst-dsp.h" +#include "sst.h" + +struct sst_machines { + char codec_id[32]; + char board[32]; + char machine[32]; + void (*machine_quirk)(void); + char firmware[32]; + struct sst_platform_info *pdata; + +}; + +/* LPE viewpoint addresses */ +#define SST_BYT_IRAM_PHY_START 0xff2c0000 +#define SST_BYT_IRAM_PHY_END 0xff2d4000 +#define SST_BYT_DRAM_PHY_START 0xff300000 +#define SST_BYT_DRAM_PHY_END 0xff320000 +#define SST_BYT_IMR_VIRT_START 0xc0000000 /* virtual addr in LPE */ +#define SST_BYT_IMR_VIRT_END 0xc01fffff +#define SST_BYT_SHIM_PHY_ADDR 0xff340000 +#define SST_BYT_MBOX_PHY_ADDR 0xff344000 +#define SST_BYT_DMA0_PHY_ADDR 0xff298000 +#define SST_BYT_DMA1_PHY_ADDR 0xff29c000 +#define SST_BYT_SSP0_PHY_ADDR 0xff2a0000 +#define SST_BYT_SSP2_PHY_ADDR 0xff2a2000 + +#define BYT_FW_MOD_TABLE_OFFSET 0x80000 +#define BYT_FW_MOD_TABLE_SIZE 0x100 +#define BYT_FW_MOD_OFFSET (BYT_FW_MOD_TABLE_OFFSET + BYT_FW_MOD_TABLE_SIZE) + +static const struct sst_info byt_fwparse_info = { + .use_elf = false, + .max_streams = 25, + .iram_start = SST_BYT_IRAM_PHY_START, + .iram_end = SST_BYT_IRAM_PHY_END, + .iram_use = true, + .dram_start = SST_BYT_DRAM_PHY_START, + .dram_end = SST_BYT_DRAM_PHY_END, + .dram_use = true, + .imr_start = SST_BYT_IMR_VIRT_START, + .imr_end = SST_BYT_IMR_VIRT_END, + .imr_use = true, + .mailbox_start = SST_BYT_MBOX_PHY_ADDR, + .num_probes = 0, + .lpe_viewpt_rqd = true, +}; + +static const struct sst_ipc_info byt_ipc_info = { + .ipc_offset = 0, + .mbox_recv_off = 0x400, +}; + +static const struct sst_lib_dnld_info byt_lib_dnld_info = { + .mod_base = SST_BYT_IMR_VIRT_START, + .mod_end = SST_BYT_IMR_VIRT_END, + .mod_table_offset = BYT_FW_MOD_TABLE_OFFSET, + .mod_table_size = BYT_FW_MOD_TABLE_SIZE, + .mod_ddr_dnld = false, +}; + +static const struct sst_res_info byt_rvp_res_info = { + .shim_offset = 0x140000, + .shim_size = 0x000100, + .shim_phy_addr = SST_BYT_SHIM_PHY_ADDR, + .ssp0_offset = 0xa0000, + .ssp0_size = 0x1000, + .dma0_offset = 0x98000, + .dma0_size = 0x4000, + .dma1_offset = 0x9c000, + .dma1_size = 0x4000, + .iram_offset = 0x0c0000, + .iram_size = 0x14000, + .dram_offset = 0x100000, + .dram_size = 0x28000, + .mbox_offset = 0x144000, + .mbox_size = 0x1000, + .acpi_lpe_res_index = 0, + .acpi_ddr_index = 2, + .acpi_ipc_irq_index = 5, +}; + +struct sst_platform_info byt_rvp_platform_data = { + .probe_data = &byt_fwparse_info, + .ipc_info = &byt_ipc_info, + .lib_info = &byt_lib_dnld_info, + .res_info = &byt_rvp_res_info, + .platform = "sst-mfld-platform", +}; + +static int sst_platform_get_resources(struct intel_sst_drv *ctx) +{ + struct resource *rsrc; + struct platform_device *pdev = to_platform_device(ctx->dev); + + /* All ACPI resource request here */ + /* Get Shim addr */ + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_lpe_res_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid SHIM base from IFWI"); + return -EIO; + } + dev_info(ctx->dev, "LPE base: %#x size:%#x", (unsigned int) rsrc->start, + (unsigned int)resource_size(rsrc)); + + ctx->iram_base = rsrc->start + ctx->pdata->res_info->iram_offset; + ctx->iram_end = ctx->iram_base + ctx->pdata->res_info->iram_size - 1; + dev_info(ctx->dev, "IRAM base: %#x", ctx->iram_base); + ctx->iram = devm_ioremap_nocache(ctx->dev, ctx->iram_base, + ctx->pdata->res_info->iram_size); + if (!ctx->iram) { + dev_err(ctx->dev, "unable to map IRAM"); + return -EIO; + } + + ctx->dram_base = rsrc->start + ctx->pdata->res_info->dram_offset; + ctx->dram_end = ctx->dram_base + ctx->pdata->res_info->dram_size - 1; + dev_info(ctx->dev, "DRAM base: %#x", ctx->dram_base); + ctx->dram = devm_ioremap_nocache(ctx->dev, ctx->dram_base, + ctx->pdata->res_info->dram_size); + if (!ctx->dram) { + dev_err(ctx->dev, "unable to map DRAM"); + return -EIO; + } + + ctx->shim_phy_add = rsrc->start + ctx->pdata->res_info->shim_offset; + dev_info(ctx->dev, "SHIM base: %#x", ctx->shim_phy_add); + ctx->shim = devm_ioremap_nocache(ctx->dev, ctx->shim_phy_add, + ctx->pdata->res_info->shim_size); + if (!ctx->shim) { + dev_err(ctx->dev, "unable to map SHIM"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->shim_phy_add = ctx->pdata->res_info->shim_phy_addr; + + /* Get mailbox addr */ + ctx->mailbox_add = rsrc->start + ctx->pdata->res_info->mbox_offset; + dev_info(ctx->dev, "Mailbox base: %#x", ctx->mailbox_add); + ctx->mailbox = devm_ioremap_nocache(ctx->dev, ctx->mailbox_add, + ctx->pdata->res_info->mbox_size); + if (!ctx->mailbox) { + dev_err(ctx->dev, "unable to map mailbox"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->mailbox_add = ctx->info.mailbox_start; + + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_ddr_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid DDR base from IFWI"); + return -EIO; + } + ctx->ddr_base = rsrc->start; + ctx->ddr_end = rsrc->end; + dev_info(ctx->dev, "DDR base: %#x", ctx->ddr_base); + ctx->ddr = devm_ioremap_nocache(ctx->dev, ctx->ddr_base, + resource_size(rsrc)); + if (!ctx->ddr) { + dev_err(ctx->dev, "unable to map DDR"); + return -EIO; + } + + /* Find the IRQ */ + ctx->irq_num = platform_get_irq(pdev, + ctx->pdata->res_info->acpi_ipc_irq_index); + return 0; +} + +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_machines *sst_acpi_find_machine( + struct sst_machines *machines) +{ + struct sst_machines *mach; + bool found = false; + + for (mach = machines; mach->codec_id; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->codec_id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + +int sst_acpi_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + int ret = 0; + struct intel_sst_drv *ctx; + const struct acpi_device_id *id; + struct sst_machines *mach; + struct platform_device *mdev; + struct platform_device *plat_dev; + unsigned int dev_id; + + id = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!id) + return -ENODEV; + dev_dbg(dev, "for %s", id->id); + + mach = (struct sst_machines *)id->driver_data; + mach = sst_acpi_find_machine(mach); + if (mach == NULL) { + dev_err(dev, "No matching machine driver found\n"); + return -ENODEV; + } + + ret = kstrtouint(id->id, 16, &dev_id); + if (ret < 0) { + dev_err(dev, "Unique device id conversion error: %d\n", ret); + return ret; + } + + dev_dbg(dev, "ACPI device id: %x\n", dev_id); + + plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0); + if (plat_dev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform); + return -ENODEV; + } + + /* Create platform device for sst machine driver */ + mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0); + if (mdev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->machine); + return -ENODEV; + } + + ret = sst_alloc_drv_context(&ctx, dev, dev_id); + if (ret < 0) + return ret; + + /* Fill sst platform data */ + ctx->pdata = mach->pdata; + strcpy(ctx->firmware_name, mach->firmware); + + ret = sst_platform_get_resources(ctx); + if (ret) + return ret; + + ret = sst_context_init(ctx); + if (ret < 0) + return ret; + + /* need to save shim registers in BYT */ + ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), + GFP_KERNEL); + if (!ctx->shim_regs64) { + return -ENOMEM; + goto do_sst_cleanup; + } + + sst_configure_runtime_pm(ctx); + platform_set_drvdata(pdev, ctx); + return ret; + +do_sst_cleanup: + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + dev_err(ctx->dev, "failed with %d\n", ret); + return ret; +} + +/** +* intel_sst_remove - remove function +* +* @pdev: platform device structure +* +* This function is called by OS when a device is unloaded +* This frees the interrupt etc +*/ +int sst_acpi_remove(struct platform_device *pdev) +{ + struct intel_sst_drv *ctx; + + ctx = platform_get_drvdata(pdev); + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct sst_machines sst_acpi_bytcr[] = { + {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + &byt_rvp_platform_data }, + {}, +}; + +static const struct acpi_device_id sst_acpi_ids[] = { + { "80860F28", (unsigned long)&sst_acpi_bytcr}, + { }, +}; + +static struct platform_driver sst_acpi_driver = { + .driver = { + .name = "intel_sst_acpi", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(sst_acpi_ids), + .pm = &intel_sst_pm, + }, + .probe = sst_acpi_probe, + .remove = sst_acpi_remove, +}; + +module_platform_driver(sst_acpi_driver); + +MODULE_DESCRIPTION("Intel (R) SST(R) Audio Engine ACPI Driver"); +MODULE_AUTHOR("Ramesh Babu K V"); +MODULE_AUTHOR("Omair Mohammed Abdullah"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("sst"); diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c index 9a5df19..4b77208 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/sst/sst_pvt.c @@ -117,6 +117,7 @@ unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: val = sst_shim_read64(sst->shim, addr); break; } @@ -128,6 +129,7 @@ void write_shim_data(struct intel_sst_drv *sst, int addr, { switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst_shim_write64(sst->shim, addr, (u64) data); break; } -- 1.7.0.4

2 1

[alsa-devel] [PATCH v2 1/3] ALSA: ice1712: remove unneeded return statement
by Sudip Mukherjee 14 Nov '14

14 Nov '14

the functions: snd_ice1712_akm4xxx_build_controls snd_ice1712_build_pro_mixer snd_ctl_add snd_ak4114_build prodigy192_ak4114_init snd_ak4113_build are all returning either 0 or a negetive error value. so we can easily remove the check for a negative value and return the value instead. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v2: returning the return value of the function directly. sound/pci/ice1712/hoontech.c | 6 +----- sound/pci/ice1712/ice1712.c | 19 ++++++------------- sound/pci/ice1712/ice1724.c | 7 ++----- sound/pci/ice1712/juli.c | 5 +---- sound/pci/ice1712/prodigy192.c | 4 +--- sound/pci/ice1712/quartet.c | 5 +---- 6 files changed, 12 insertions(+), 34 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 59e37c5..a40001c 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -309,11 +309,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice) return err; /* ak4524 controls */ - err = snd_ice1712_akm4xxx_build_controls(ice); - if (err < 0) - return err; - - return 0; + return snd_ice1712_akm4xxx_build_controls(ice); } static int snd_ice1712_ez8_init(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 48a0c33..5975334 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1295,10 +1295,7 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd return err; } - err = snd_ice1712_build_pro_mixer(ice); - if (err < 0) - return err; - return 0; + return snd_ice1712_build_pro_mixer(ice); } /* @@ -1545,10 +1542,9 @@ static int snd_ice1712_ac97_mixer(struct snd_ice1712 *ice) dev_warn(ice->card->dev, "cannot initialize ac97 for consumer, skipped\n"); else { - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice)); - if (err < 0) - return err; - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, + ice)); } } @@ -2497,11 +2493,8 @@ static int snd_ice1712_build_controls(struct snd_ice1712 *ice) err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_volume_rate, ice)); if (err < 0) return err; - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); } static int snd_ice1712_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f633e3b..ea53167 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2497,11 +2497,8 @@ static int snd_vt1724_build_controls(struct snd_ice1712 *ice) return err; } - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); } static int snd_vt1724_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 7a6c078..a1536c1 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -475,11 +475,8 @@ static int juli_add_controls(struct snd_ice1712 *ice) return err; /* only capture SPDIF over AK4114 */ - err = snd_ak4114_build(spec->ak4114, NULL, + return snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } /* diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 1eb151aa..3919aed 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -758,10 +758,8 @@ static int prodigy192_init(struct snd_ice1712 *ice) "AK4114 initialized with status %d\n", err); } else dev_dbg(ice->card->dev, "AK4114 not found\n"); - if (err < 0) - return err; - return 0; + return err; } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d4caf9d..6f55e02 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -833,11 +833,8 @@ static int qtet_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; /* only capture SPDIF over AK4113 */ - err = snd_ak4113_build(spec->ak4113, + return snd_ak4113_build(spec->ak4113, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) -- 1.8.1.2

2 7

[alsa-devel] [PATCH v3] ASoC: add TI ts3a227e headset chip driver
by Dylan Reid 14 Nov '14

14 Nov '14

The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on the jack as buttons 0-3 mapped to KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND. This will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v2: Switch from codec to component driver Changes since v1: Use button to key mapping instead of separate input device --- .../devicetree/bindings/sound/ts3a227e.txt | 26 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ts3a227e.c | 314 +++++++++++++++++++++ sound/soc/codecs/ts3a227e.h | 17 ++ 5 files changed, 364 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..1d12057 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,314 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + bool plugged; + bool mic_present; + unsigned int buttons_held; +}; + +/* Button values to be reported on the jack */ +static const int ts3a227e_buttons[] = { + SND_JACK_BTN_0, + SND_JACK_BTN_1, + SND_JACK_BTN_2, + SND_JACK_BTN_3, +}; + +#define TS3A227E_NUM_BUTTONS 4 +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_MICROPHONE | \ + SND_JACK_BTN_0 | \ + SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | \ + SND_JACK_BTN_3) + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + unsigned int i; + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (ts3a227e->buttons_held & (1 << i)) + report |= ts3a227e_buttons[i]; + } + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + ts3a227e->buttons_held = 0; + if (mic_present) { + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, + TS3A227E_REG_SETTING_2, + KP_ENABLE, KP_ENABLE); + } + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + ts3a227e->buttons_held |= (1 << i); + if (kp_int_reg & RELEASE_MASK(i)) + ts3a227e->buttons_held &= ~(1 << i); + } + + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events 0-3 will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_component_driver ts3a227e_soc_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (ts3a227e == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = devm_snd_soc_register_component(&i2c->dev, &ts3a227e_soc_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..e2acf9c --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + +#endif -- 2.1.2.330.g565301e

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Modify the default value and add it to patch table
by Oder Chiou 14 Nov '14

14 Nov '14

Modify the default value and add it to patch table Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..9ae2e84 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,7 +55,8 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, {RT5677_PR_BASE + 0x13, 0x0312}, {RT5677_PR_BASE + 0x1e, 0x0000}, @@ -173,7 +174,7 @@ static const struct reg_default rt5677_reg[] = { {RT5677_ASRC_9 , 0x0000}, {RT5677_ASRC_10 , 0x0000}, {RT5677_ASRC_11 , 0x0000}, - {RT5677_ASRC_12 , 0x0008}, + {RT5677_ASRC_12 , 0x0018}, {RT5677_ASRC_13 , 0x0000}, {RT5677_ASRC_14 , 0x0000}, {RT5677_ASRC_15 , 0x0000}, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5645: two jacks for hp and mic
by Bard Liao 14 Nov '14

14 Nov '14

Some OS need headphone and microphone to be separated. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 17 ++++++++--------- sound/soc/codecs/rt5645.h | 5 +++-- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 82d705a..31b5b1a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2203,8 +2203,7 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int rt5645_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack) +static int rt5645_jack_detect(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); int gpio_state, jack_type = 0; @@ -2247,19 +2246,19 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, snd_soc_dapm_sync(&codec->dapm); } - snd_soc_jack_report(rt5645->jack, jack_type, SND_JACK_HEADSET); - + snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); + snd_soc_jack_report(rt5645->mic_jack, jack_type, SND_JACK_MICROPHONE); return 0; } int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack) + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - rt5645->jack = jack; - - rt5645_jack_detect(codec, rt5645->jack); + rt5645->hp_jack = hp_jack; + rt5645->mic_jack = mic_jack; + rt5645_jack_detect(codec); return 0; } @@ -2270,7 +2269,7 @@ static void rt5645_jack_detect_work(struct work_struct *work) struct rt5645_priv *rt5645 = container_of(work, struct rt5645_priv, jack_detect_work.work); - rt5645_jack_detect(rt5645->codec, rt5645->jack); + rt5645_jack_detect(rt5645->codec); } static irqreturn_t rt5645_irq(int irq, void *data) diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 196daf0..c72220a 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2173,7 +2173,8 @@ struct rt5645_priv { struct rt5645_platform_data pdata; struct regmap *regmap; struct i2c_client *i2c; - struct snd_soc_jack *jack; + struct snd_soc_jack *hp_jack; + struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; int sysclk; @@ -2188,6 +2189,6 @@ struct rt5645_priv { }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, - struct snd_soc_jack *jack); + struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); #endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix SMALL_POP bit definition
by Fabio Estevam 14 Nov '14

14 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> On a mx28evk with a sgtl5000 codec we notice a loud 'click' sound to happen 5 seconds after the end of a playback. The SMALL_POP bit should fix this, but its definition is incorrect: according to the sgtl5000 manual it is bit 0 of CHIP_REF_CTRL register, not bit 1. Fix the definition accordingly and enable the bit as intended per the code comment. After applying this change, no loud 'click' sound is heard after playback Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/sgtl5000.h | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6bb77d7..dab9b15 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1299,8 +1299,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) /* enable small pop, introduce 400ms delay in turning off */ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - SGTL5000_SMALL_POP, - SGTL5000_SMALL_POP); + SGTL5000_SMALL_POP, 1); /* disable short cut detector */ snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 2f8c889..bd7a344 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -275,7 +275,7 @@ #define SGTL5000_BIAS_CTRL_MASK 0x000e #define SGTL5000_BIAS_CTRL_SHIFT 1 #define SGTL5000_BIAS_CTRL_WIDTH 3 -#define SGTL5000_SMALL_POP 0x0001 +#define SGTL5000_SMALL_POP 0 /* * SGTL5000_CHIP_MIC_CTRL -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rt5670 : Add ACPI match ID for Intel CHT/BSW platforms
by mengdong.lin＠intel.com 14 Nov '14

14 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds the ACPI match ID for rt5670/5672 codec. So on Intel CherryTrail/Braswell platforms, the codec can be enumerated from ACPI and depends on ACPI to get platform-specific info and power saving. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c4c42d5..fd32501 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -16,6 +16,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/acpi.h> #include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -2503,6 +2504,14 @@ static const struct i2c_device_id rt5670_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5670_acpi_match[] = { + { "10EC5670", 0}, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); +#endif + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2691,6 +2700,7 @@ static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5670_acpi_match), }, .probe = rt5670_i2c_probe, .remove = rt5670_i2c_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: ice1712: remove unneeded return statement
by Sudip Mukherjee 14 Nov '14

14 Nov '14

the functions: snd_ice1712_akm4xxx_build_controls snd_ice1712_build_pro_mixer snd_ctl_add snd_ak4114_build prodigy192_ak4114_init snd_ak4113_build are all returning either 0 or a negetive error value. so we can easily remove the check for a negative value and return the value instead. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- i have not given Jaroslav Kysela <perex(a)perex.cz> in the To list, as in my last mail, all mails bounced with the error: "cant create user output file" sound/pci/ice1712/hoontech.c | 5 +---- sound/pci/ice1712/ice1712.c | 13 +++---------- sound/pci/ice1712/ice1724.c | 5 +---- sound/pci/ice1712/juli.c | 4 +--- sound/pci/ice1712/prodigy192.c | 4 +--- sound/pci/ice1712/quartet.c | 4 +--- 6 files changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 59e37c5..2c53154 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -310,10 +310,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice) /* ak4524 controls */ err = snd_ice1712_akm4xxx_build_controls(ice); - if (err < 0) - return err; - - return 0; + return err; } static int snd_ice1712_ez8_init(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 48a0c33..b97a0f9 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1296,9 +1296,7 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd } err = snd_ice1712_build_pro_mixer(ice); - if (err < 0) - return err; - return 0; + return err; } /* @@ -1546,9 +1544,7 @@ static int snd_ice1712_ac97_mixer(struct snd_ice1712 *ice) "cannot initialize ac97 for consumer, skipped\n"); else { err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice)); - if (err < 0) - return err; - return 0; + return err; } } @@ -2498,10 +2494,7 @@ static int snd_ice1712_build_controls(struct snd_ice1712 *ice) if (err < 0) return err; err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return err; } static int snd_ice1712_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f633e3b..fa88485 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2498,10 +2498,7 @@ static int snd_vt1724_build_controls(struct snd_ice1712 *ice) } err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return err; } static int snd_vt1724_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 7a6c078..d6f2fb0 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -477,9 +477,7 @@ static int juli_add_controls(struct snd_ice1712 *ice) /* only capture SPDIF over AK4114 */ err = snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; + return err; } /* diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 1eb151aa..3919aed 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -758,10 +758,8 @@ static int prodigy192_init(struct snd_ice1712 *ice) "AK4114 initialized with status %d\n", err); } else dev_dbg(ice->card->dev, "AK4114 not found\n"); - if (err < 0) - return err; - return 0; + return err; } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d4caf9d..3903e05 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -835,9 +835,7 @@ static int qtet_add_controls(struct snd_ice1712 *ice) /* only capture SPDIF over AK4113 */ err = snd_ak4113_build(spec->ak4113, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; + return err; } static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) -- 1.8.1.2

2 4

[alsa-devel] [PATCH] 6fire: Convert byte_rev_table uses to bitrev8
by Wang, Yalin 14 Nov '14

14 Nov '14

Use the inline function instead of directly indexing the array. This allows some architectures with hardware instructions for bit reversals to eliminate the array. Signed-off-by: Joe Perches <joe(a)perches.com> Signed-off-by: Yalin Wang <yalin.wang(a)sonymobile.com> --- sound/usb/6fire/firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 3b02e54..2e39c3f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -316,7 +316,7 @@ static int usb6fire_fw_fpga_upload( while (c != end) { for (i = 0; c != end && i < FPGA_BUFSIZE; i++, c++) - buffer[i] = byte_rev_table[(u8) *c]; + buffer[i] = bitrev8((u8) *c); ret = usb6fire_fw_fpga_write(device, buffer, i); if (ret < 0) { -- 2.1.1

2 1

[alsa-devel] Low sample rates for SGTL5000 on i.MX28
by Craig McQueen 14 Nov '14

14 Nov '14

I'm testing the 3.18 kernel for i.MX28 EVK, which has an SGTL5000 CODEC. I notice that audio cannot be output at 8 kHz or 11.025 kHz, although the SGTL5000 data sheet table 8 indicates this should be possible. If I try to play or record at 8 kHz or 11.025 kHz, I get an error message: # aplay test_8k.wav [ 6168.910201] mxs_evk_startup: Skipped programming wolfson codec Playing WAVE 'test_8k.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono [ 6168.941211] sgtl5000 0-000a: Invalid mclk frequency: 4.096MHz [ 6168.947047] mxs-sgtl5000 sound.11: ASoC: machine hw_params failed: -22 aplay: set_params:1145: Unable to install hw params: ... For an 8 kHz sample rate, I get Invalid mclk frequency: 4.096MHz For 11.025 kHz sample rate, I get Invalid mclk frequency: 5.644MHz Looking in the code (/sound/soc/mxs/mxs-sgtl5000c), I see: /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ if (mclk < 8000000 || mclk > 27000000) { dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n", mclk / 1000000, mclk / 1000 % 1000); return -EINVAL; } But a sysclk constraint of 8 to 27 MHz only applies if the supplied clock is asynchronous to the frame clock, and the internal clock is being generated by PLL (which it's not in this case). So I tried removing this constraint. But I found that from codecs/sgtl5000.c, I would get the error "PLL not supported in slave mode". I tried changing the switch (sgtl5000->sysclk / sys_fs) to switch (sgtl5000->sysclk / frame_rate), on the basis of the SGTL5000 data sheet table 8. But the resulting audio sounds funny (frequency shifted). Looking in the SGTL5000 data sheet table 18, it seems unclear whether the MCLK_FREQ field is a multiple of the sample rate (Fs) or the internal system sample rate SYS_FS. It seems that there is some inconsistency in the interpretation of this in codecs/sgtl5000.c compared to mxs/mxs-sgtl5000.c. What is the correct interpretation of the SGTL5000 data sheet MCLK_FREQ field, and can the mxs-sgtl5000 driver be fixed to support 8 kHz and 11.025 kHz sample rates? -- Craig McQueen

2 6

[alsa-devel] Stale data at start of audio recording on Freescale i.MX28 EVK with SGTL5000
by Craig McQueen 13 Nov '14

13 Nov '14

I was testing the 3.14.19 kernel on Freescale i.MX28 EVK. Since then, I've retested and confirmed this also on 3.18-rc4 kernel. I am doing audio recordings through the SGTL5000 driver as such: arecord -D hw:0,1 -f s16 -c 2 -r 16000 -d 5 test.wav I'm finding that the first 400-500 ms of the saved data is actually stale data from the end of the previous recording. To see this, do the following: 1. Provide some test audio input in the LINE INPUT jack. E.g. a 100 Hz sine wave, or some music. 2. Record audio with e.g. arecord -D hw:0,1 -f s16 -c 2 -r 16000 -d 5 test1.wav 3. Change the test audio input, e.g. a 150 Hz sine wave, or some different music. 4. Record a new audio file with e.g. arecord -D hw:0,1 -f s16 -c 2 -r 16000 -d 5 test2.wav Observe that the first 400-500 ms of test2.wav contains stale data from the end of the previous recording. -- Craig McQueen

1 0

[alsa-devel] [PATCH v2] ASoC: add TI ts3a227e headset chip driver
by Dylan Reid 13 Nov '14

13 Nov '14

The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on the jack as buttons 0-3 mapped to KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND. This will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Use button to key mapping instead of separate input device --- .../devicetree/bindings/sound/ts3a227e.txt | 26 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ts3a227e.c | 321 +++++++++++++++++++++ sound/soc/codecs/ts3a227e.h | 17 ++ 5 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..4ca6ddc --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,321 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + bool plugged; + bool mic_present; + unsigned int buttons_held; +}; + +/* Button values to be reported on the jack */ +static const int ts3a227e_buttons[] = { + SND_JACK_BTN_0, + SND_JACK_BTN_1, + SND_JACK_BTN_2, + SND_JACK_BTN_3, +}; + +#define TS3A227E_NUM_BUTTONS 4 +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_MICROPHONE | \ + SND_JACK_BTN_0 | \ + SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | \ + SND_JACK_BTN_3) + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + unsigned int i; + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (ts3a227e->buttons_held & (1 << i)) + report |= ts3a227e_buttons[i]; + } + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + ts3a227e->buttons_held = 0; + if (mic_present) { + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, + TS3A227E_REG_SETTING_2, + KP_ENABLE, KP_ENABLE); + } + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + ts3a227e->buttons_held |= (1 << i); + if (kp_int_reg & RELEASE_MASK(i)) + ts3a227e->buttons_held &= ~(1 << i); + } + + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @codec: codec to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events 0-3 will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_codec_get_drvdata(codec); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_codec_driver ts3a227e_codec_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (ts3a227e == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &ts3a227e_codec_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static int ts3a227e_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .remove = ts3a227e_i2c_remove, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..8a58886 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 2.1.2.330.g565301e

2 2

[alsa-devel] Problems with usb soundcard in kernels > 3.3
by Bogdan Veringioiu 13 Nov '14

13 Nov '14

Hello all, After spending a couple of days searching the web for a solution, I write here to ask for help. I am using an usb soundcard (Phoenix Audio Technologies Phnx MT201solo) on Debian Wheezy, and started to experience problems when using kernels > 3.3. The issue I am experiencing is a ~4 seconds delay in playing sound. I am reproducing the problem using aplay. The sound play is being delayed by more or less 4 seconds from the moment the file is sent to the card. I have tested with kernels 3.2 and 3.3, which work properly. I have noticed this delay/wait when using kernels 3.15, 3.16. Below are some details: WORKING PROPERLY # uname -rv 3.2.0-4-686-pae #1 SMP Debian 3.2.63-2 # dpkg -l | grep alsa ii alsa-base 1.0.25+3~deb7u1 all ALSA driver configuration files ii alsa-utils 1.0.25-4 i386 Utilities for configuring and using ALSA # dmesg .... usb 3-1: New USB device found, idVendor=1de7, idProduct=0011 usb 3-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 3-1: Product: Phnx MT201solo usb 3-1: Manufacturer: Phoenix Audio Technologies .... # time aplay /usr/share/sounds/alsa/Rear_Left.wav Playing WAVE '/usr/share/sounds/alsa/Rear_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono real 0m2.498s <--- this is OK user 0m0.032s sys 0m0.024s DELAY INTRODUCED # uname -rv 3.16-0.bpo.2-686-pae #1 SMP Debian 3.16.3-2~bpo70+1 (2014-09-21) # dpkg -l | grep alsa ii alsa-base 1.0.25+3~deb7u1 all ALSA driver configuration files ii alsa-utils 1.0.25-4 i386 Utilities for configuring and using ALSA # dmesg .... usb 2-1: New USB device found, idVendor=1de7, idProduct=0011 usb 2-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 2-1: Product: Phnx MT201solo usb 2-1: Manufacturer: Phoenix Audio Technologies .... # time aplay /usr/share/sounds/alsa/Rear_Left.wav Playing WAVE '/usr/share/sounds/alsa/Rear_Left.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono real 0m6.387s <----4 more seconds delay, NOK user 0m0.016s sys 0m0.024s I have done a strace, see below where the 4 seconds are lost, maybe it helps: open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK|O_CLOEXEC) = 4 fcntl64(4, F_SETFD, FD_CLOEXEC) = 0 ioctl(4, AGPIOC_ACQUIRE or APM_IOC_STANDBY, 0xbf823f4c) = 0 close(3) = 0 ioctl(4, AGPIOC_ACQUIRE or APM_IOC_STANDBY, 0xbf823dac) = 0 fcntl64(4, F_GETFL) = 0x802 (flags O_RDWR|O_NONBLOCK) ioctl(4, AGPIOC_INFO, 0xbf823d14) = 0 clock_gettime(CLOCK_MONOTONIC, {5839, 700325532}) = 0 ioctl(4, AGPIOC_SETUP, 0xbf823d1c) = 0 mmap2(NULL, 4096, PROT_READ, MAP_SHARED, 4, 0x80000) = 0xb72db000 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0x81000) = 0xb72da000 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4110, 0xbf823ee0) = 0 ioctl(4, 0xc25c4111, 0xbf823ee0) = 0 ioctl(4, 0xc0684113, 0xbf823d94) = 0 ioctl(4, 0x80104132, 0xbf823cd0) = 0 ioctl(4, 0x80104132, 0xbf823cd0) = 0 mmap2(NULL, 65536, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0) = 0xb7293000 ----> ioctl(4, 0x4140, 0xb7718ff4) = 0 <---- !!! Here is where the 4 seconds are lost !!! ioctl(4, 0xc0684113, 0xbf823e60) = 0 ioctl(4, 0x4142, 0xb7709c16) = 0 ioctl(4, AGPIOC_INFO, 0xbf8241a0) = 0 shmget(0x56a4d6, 131072, IPC_CREAT|0660) = 163841 shmctl(163841, IPC_64|IPC_STAT, 0xbf82415c) = 0 shmctl(163841, IPC_64|IPC_SET, 0xbf82415c) = 0 .... I think this is a problem in the snd-usb-audio module introduced in kernels > 3.3. Can anyone help me with this issue? Thanks, Bogdan

2 5

[alsa-devel] [PATCH v2] Scarlett mixer interface inclusion
by Chris J Arges 13 Nov '14

13 Nov '14

This is a followup to David Henningsson's PATCH [1] to include the work of Tobias Hoffman and Robin Gareus in getting the Scarlett mixer interface working. I've read through the comments and tried to include fixes where appropriate. Overall, I'd like to get a good base commit and start working on other commits that address any major structual changes such as making functions more generic or any other cleanup. I've tested this on my Scarlett 18i8 mixer and it does work, however there are still a few issues which need to be fixed: 1) When loading a device one gets the following messages: snd-usb-audio 1-1.1:1.0: control 2:0:0:Master Playback Switch:0 is already present snd-usb-audio: probe of 1-1.1:1.3 failed with error -5 2) When unloading a device there are numerous sysfs_remove_group issues: usb 1-1.1: USB disconnect, device number 6 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 54 at /build/buildd/linux-3.16.0/fs/sysfs/group.c:219 sysfs_remove_group+0x99/0xa0() sysfs group ffffffff82cbd6e0 not found for kobject 'midi1' Modules linked in: snd_usb_audio(OE) snd_usbmidi_lib(OE) ctr ccm xt_conntrack ipt_REJECT xt_CHECKSUM iptable_mangle ipt_MASQUERADE iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack xt_tcpudp bridge stp llc iptable_filter ip_tables x_tables arc4 iwldvm snd_hda_codec_hdmi mac80211 snd_hda_codec_conexant snd_hda_codec_generic iwlwifi snd_hda_intel uvcvideo snd_hda_controller videobuf2_vmalloc snd_hda_codec videobuf2_memops videobuf2_core snd_hwdep v4l2_common snd_pcm intel_rapl x86_pkg_temp_thermal videodev intel_powerclamp media coretemp kvm_intel bnep kvm btusb rfcomm cfg80211 joydev bluetooth serio_raw 6lowpan_iphc thinkpad_acpi nvram snd_seq_midi lpc_ich snd_seq_midi_event snd_rawmidi shpchp snd_seq mei_me snd_seq_device snd_timer mei snd soundcore parport_pc ppdev mac_hid lp parport binfmt_misc nls_iso8859_1 dm_crypt crct10dif_pclmul crc32_pclmul ghash_clmulni_intel aesni_intel aes_x86_64 lrw gf128mul glue_helper ablk_helper cryptd i915 psmouse ahci libahci firewire_ohci sdhci_pci firewire_core sdhci crc_itu_t i2c_algo_bit e1000e drm_kms_helper drm ptp pps_core wmi video CPU: 0 PID: 54 Comm: khubd Tainted: G OE 3.16.0-23-generic #30-Ubuntu Hardware name: LENOVO 4177CTO/4177CTO, BIOS 83ET76WW (1.46 ) 07/05/2013 0000000000000009 ffff880407b4f920 ffffffff8277fcbc ffff880407b4f968 ffff880407b4f958 ffffffff8206fd8d 0000000000000000 ffffffff82cbd6e0 ffff880405c7b410 0000000000000001 ffff8803eb059cf8 ffff880407b4f9b8 Call Trace: [<ffffffff8277fcbc>] dump_stack+0x45/0x56 [<ffffffff8206fd8d>] warn_slowpath_common+0x7d/0xa0 [<ffffffff8206fdfc>] warn_slowpath_fmt+0x4c/0x50 [<ffffffff82256538>] ? kernfs_find_and_get_ns+0x48/0x60 [<ffffffff8225a259>] sysfs_remove_group+0x99/0xa0 [<ffffffff824d6927>] dpm_sysfs_remove+0x57/0x60 [<ffffffff824cb625>] device_del+0x45/0x1d0 [<ffffffff824cb7ce>] device_unregister+0x1e/0x70 [<ffffffff824cb89c>] device_destroy+0x3c/0x50 [<ffffffffc045f34b>] sound_remove_unit+0x5b/0x90 [soundcore] [<ffffffffc045f3aa>] unregister_sound_special+0x2a/0x30 [soundcore] [<ffffffffc04cc9b0>] snd_unregister_oss_device+0x60/0x100 [snd] [<ffffffffc0550ca9>] snd_rawmidi_dev_disconnect+0x119/0x160 [snd_rawmidi] [<ffffffffc04cc14c>] __snd_device_disconnect.part.1+0x1c/0x60 [snd] [<ffffffffc04cc437>] snd_device_disconnect_all+0x47/0x60 [snd] [<ffffffffc04c59f7>] snd_card_disconnect+0x147/0x1f0 [snd] [<ffffffffc09cf5a8>] usb_audio_disconnect+0x78/0x1a0 [snd_usb_audio] [<ffffffff825867b4>] usb_unbind_interface+0x74/0x2b0 [<ffffffff824cf76f>] __device_release_driver+0x7f/0xf0 [<ffffffff824cf803>] device_release_driver+0x23/0x30 [<ffffffff824cf0f8>] bus_remove_device+0x108/0x180 [<ffffffff824cb709>] device_del+0x129/0x1d0 [<ffffffff82583f21>] usb_disable_device+0x91/0x290 [<ffffffff82578688>] usb_disconnect+0x98/0x2f0 [<ffffffff8257a6cf>] hub_port_connect+0x7f/0xac0 [<ffffffff8257b9a3>] hub_events+0x893/0xda0 [<ffffffff8257beed>] hub_thread+0x3d/0x160 [<ffffffff820b9590>] ? prepare_to_wait_event+0x100/0x100 [<ffffffff8257beb0>] ? hub_events+0xda0/0xda0 [<ffffffff82094aeb>] kthread+0xdb/0x100 [<ffffffff82094a10>] ? kthread_create_on_node+0x1c0/0x1c0 [<ffffffff82787c3c>] ret_from_fork+0x7c/0xb0 [<ffffffff82094a10>] ? kthread_create_on_node+0x1c0/0x1c0 ---[ end trace 7a8cdc4a9c65ddf5 ]--- However this issue occured even without the patch applied. Please follow up with additional comments and review and I'll be happy to incorporate them into the next patch version. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/081986.ht… Chris J Arges (1): Scarlett mixer interface for 6i6, 18i6, 18i8 and 18i20 sound/usb/Makefile | 1 + sound/usb/mixer.c | 27 +- sound/usb/quirks-table.h | 51 -- sound/usb/scarlett/scarlettmixer.c | 1264 ++++++++++++++++++++++++++++++++++++ sound/usb/scarlett/scarlettmixer.h | 6 + 5 files changed, 1293 insertions(+), 56 deletions(-) create mode 100644 sound/usb/scarlett/scarlettmixer.c create mode 100644 sound/usb/scarlett/scarlettmixer.h -- 2.1.0

6 59

[alsa-devel] [PATCH v2] ASoC: rt5645: add register setting for TDM
by Bard Liao 13 Nov '14

13 Nov '14

We need to set extra register to avoid a recording issue in TDM mode. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Change regmap_update_bits(rt5645->regmap to snd_soc_update_bits(codec since rt5645 struct is not in rt5645_set_tdm_slot --- sound/soc/codecs/rt5645.c | 5 ++++- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d6298b3..82d705a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2114,8 +2114,11 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, struct snd_soc_codec *codec = dai->codec; unsigned int val = 0; - if (rx_mask || tx_mask) + if (rx_mask || tx_mask) { val |= (1 << 14); + snd_soc_update_bits(codec, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + } switch (slots) { case 4: diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 82f681b..196daf0 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1855,6 +1855,7 @@ #define RT5645_M_BB_HPF_R_SFT 6 #define RT5645_G_BB_BST_MASK (0x3f) #define RT5645_G_BB_BST_SFT 0 +#define RT5645_G_BB_BST_25DB 0x14 /* MP3 Plus Control 1 (0xd0) */ #define RT5645_M_MP3_L_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/4 V2] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 13 Nov '14

13 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Changes done: - Removed ifdef for of_device_id structure as commented by Mark Brown - Supporting both variants of device name:alc5631 and rt5631 as commented by Mark Brown - Added "return -EINVAL" if cpu_of_node or codec_of_node is not found in dtb, as commented by Anil Kumar Krishna Mohan Dani (4): ASoC: ALC5631/RT5631: Add device tree binding documentation ASoC: Samsung: Add arndale_rt5631 machine driver Sound: Kconfig: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 48 ++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 11 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 166 ++++++++++++++++++++ 6 files changed, 235 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

3 8

[alsa-devel] [PATCH] ALSA: hda - Add mute LED control for Lenovo Ideapad Z560
by Takashi Iwai 13 Nov '14

13 Nov '14

Lenovo Ideapad Z560 has a mute LED that is controlled via EAPD pin 0x1b on CX20585 codec. (EAPD bit on corresponds to mute LED on.) The machine doesn't need other EAPD, so the fixup concentrates on controlling EAPD 0x1b following the vmaster state (but inversely). Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=665315 Reported-by: Szymon Kowalczyk <fazerxlo(a)o2.pl> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71e4bad06345..e9ebc7bd752c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -43,6 +43,7 @@ struct conexant_spec { unsigned int num_eapds; hda_nid_t eapds[4]; bool dynamic_eapd; + hda_nid_t mute_led_eapd; unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ @@ -163,6 +164,17 @@ static void cx_auto_vmaster_hook(void *private_data, int enabled) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, enabled); } +/* turn on/off EAPD according to Master switch (inversely!) for mute LED */ +static void cx_auto_vmaster_hook_mute_led(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct conexant_spec *spec = codec->spec; + + snd_hda_codec_write(codec, spec->mute_led_eapd, 0, + AC_VERB_SET_EAPD_BTLENABLE, + enabled ? 0x00 : 0x02); +} + static int cx_auto_build_controls(struct hda_codec *codec) { int err; @@ -223,6 +235,7 @@ enum { CXT_FIXUP_TOSHIBA_P105, CXT_FIXUP_HP_530, CXT_FIXUP_CAP_MIX_AMP_5047, + CXT_FIXUP_MUTE_LED_EAPD, }; /* for hda_fixup_thinkpad_acpi() */ @@ -557,6 +570,18 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec, } } +static void cxt_fixup_mute_led_eapd(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_eapd = 0x1b; + spec->dynamic_eapd = 1; + spec->gen.vmaster_mute.hook = cx_auto_vmaster_hook_mute_led; + } +} + /* * Fix max input level on mixer widget to 0dB * (originally it has 0x2b steps with 0dB offset 0x14) @@ -705,6 +730,10 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_cap_mix_amp_5047, }, + [CXT_FIXUP_MUTE_LED_EAPD] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_mute_led_eapd, + }, }; static const struct snd_pci_quirk cxt5045_fixups[] = { @@ -762,6 +791,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), @@ -780,6 +810,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" }, { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, + { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, {} }; -- 2.1.3

1 0

[alsa-devel] possible snd_hda_intel jack-sensing issue with device 8086:1c20
by Chris Pemberton 13 Nov '14

13 Nov '14

CentOS 7 x86_64 Asus P8Z68-V motherboard Realtek ALC892 Gnome 3 desktop nVidia GTX 770 with nVidia's driver With BIOS set to HD for both rear and front jacks, I experience the problems I reported here: https://bugs.freedesktop.org/show_bug.cgi?id=86158 Basically, sound works, but the selected output keeps auto-magically flipping from Analog-Out to Headphones and back so fast you barely hear a click.. Tried all sorts of snd_hda_intel module options and it didn't change. Finally changed BIOS to set front jacks to AC97 (left rear jacks as HD) and sound works great... but no jack sensing occurs at all... I can live without the jack sensing... but if anyone wants a stab at a fix, I'm game. I've got the HDA Analyzer downloaded and running if that helps. My device has id 8086:1c20 Perhaps similar to this bug that was brought to my attention? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Output of alsa-info.sh is here: http://www.alsa-project.org/db/?f=8cf116b0bc230524bcfa7285b2adb698078d4224 Thanks.

2 3

Re: [alsa-devel] Compressed audio support over HDMI
by Andrej Falout 13 Nov '14

13 Nov '14

> On 11/4/14, 3:40 PM, D K wrote: > > Hi, > > I recently came across the compressed audio framework used with > ALSA-SOC. > > Can the framework be used to send compressed audio to HDMI driver?.Like > > using tinycompress from usespace . > > thanks, > > HDMI already supports IEC modes when you can send AC3, DTS, etc payloads > without decoding. All you need to do is packetize the data in userspace > and send the result over the HDMI PCM device (and set some control bits > to help the receiver lock faster). > > Apologies for jumping in on this subject, I'm wondering if anyone knows if this method is of sending audio over HDMI is aplocable to DSD streams too? The reasoning being that once audio is compressed, and encoded (AC3, DTS, etc), there is not much inherently "PCM" about it. It only becomes PCM again after it is decoded and uncompressed, right? In short, it is simply a bitstream, packetized and with "some control bits set". Which is exactly what I see in HDMI specs in regards to DSD. So, does anyone know of a reason why this same method wont work for sending DSD bitstream over HDMI? Any advice very much appreciated! Kind regards, Andrej Falout

1 0

[alsa-devel] [PATCH] ASoC: rt5645: add register setting for TDM
by Bard Liao 12 Nov '14

12 Nov '14

We need to set extra register to avoid a recording issue in TDM mode. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 5 ++++- sound/soc/codecs/rt5645.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d6298b3..ac7a52a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2114,8 +2114,11 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, struct snd_soc_codec *codec = dai->codec; unsigned int val = 0; - if (rx_mask || tx_mask) + if (rx_mask || tx_mask) { val |= (1 << 14); + regmap_update_bits(rt5645->regmap, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + } switch (slots) { case 4: diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 82f681b..196daf0 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1855,6 +1855,7 @@ #define RT5645_M_BB_HPF_R_SFT 6 #define RT5645_G_BB_BST_MASK (0x3f) #define RT5645_G_BB_BST_SFT 0 +#define RT5645_G_BB_BST_25DB 0x14 /* MP3 Plus Control 1 (0xd0) */ #define RT5645_M_MP3_L_MASK (0x1 << 15) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/5] ARM: shmobile: Sound binding updates
by Geert Uytterhoeven 12 Nov '14

12 Nov '14

Hi Mark, Liam, Simon, Magnus, Morimoto-san, THis patch series add SoC-specific binding to the shmobile sound bindings, and updates/cleans up the affected dtsi files. Patches 1 and 3 are meant for the ASoC tree. Patches 2, 4, 5 and are meant for the shmobile tree. Thanks! Geert Uytterhoeven (5): ASoC: sh: fsi: Document SoC-specific bindings ARM: shmobile: sh73a0 dtsi: Add SoC-specific FSI2 compatible property ASoC: rsnd: Document SoC-specific bindings ARM: shmobile: r8a7790 dtsi: Drop "renesas,rcar_sound" compatible value ARM: shmobile: r8a7791 dtsi: Drop "renesas,rcar_sound" compatible value Documentation/devicetree/bindings/sound/renesas,fsi.txt | 9 +++++++-- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 10 +++++++--- arch/arm/boot/dts/r8a7790.dtsi | 2 +- arch/arm/boot/dts/r8a7791.dtsi | 2 +- arch/arm/boot/dts/sh73a0.dtsi | 2 +- 5 files changed, 17 insertions(+), 8 deletions(-) -- 1.9.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

3 9

[alsa-devel] [PATCH] ASoC: add TI ts3a227e headset chip driver
by Dylan Reid 12 Nov '14

12 Nov '14

The TS3A227E is an autonomous audio accessory detection and configuration switch that detects 3-pole or 4-pole audio accessories and configures internal switches to route the signals accordingly. This chip also has built-in support for the new button standard described in the Android "Wired audio headset specification" v1.0. These buttons will be reported on a separate input device. The CODEC will be added as an aux_dev and have the jack passed in from the machine driver. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- .../devicetree/bindings/sound/ts3a227e.txt | 26 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ts3a227e.c | 330 +++++++++++++++++++++ sound/soc/codecs/ts3a227e.h | 17 ++ 5 files changed, 380 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt new file mode 100644 index 0000000..e8bf23e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -0,0 +1,26 @@ +Texas Instruments TS3A227E +Autonomous Audio Accessory Detection and Configuration Switch + +The TS3A227E detect headsets of 3-ring and 4-ring standards and +switches automatically to route the microphone correctly. It also +handles key press detection in accordance with the Android audio +headset specification v1.0. + +Required properties: + + - compatible: Should contain "ti,ts3a227e". + - reg: The i2c address. Should contain <0x3b>. + - interrupt-parent: The parent interrupt controller + - interrupts: Interrupt number for /INT pin from the 227e + + +Examples: + + i2c { + ts3a227e@3b { + compatible = "ti,ts3a227e"; + reg = <0x3b>; + interrupt-parent = <&gpio>; + interrupts = <3 IRQ_TYPE_LEVEL_LOW>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a68d173..243ec86 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C + select SND_SOC_TS3A227E if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X @@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X config SND_SOC_TLV320DAC33 tristate +config SND_SOC_TS3A227E + tristate "TI Headset/Mic detect and keypress chip" + depends on I2C + config SND_SOC_TWL4030 select MFD_TWL4030_AUDIO tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5dce451..a1eb7ef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o +snd-soc-ts3a227e-objs := ts3a227e.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o @@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o +obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c new file mode 100644 index 0000000..c1aa2f5 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.c @@ -0,0 +1,330 @@ +/* + * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/soc.h> + +struct ts3a227e { + struct regmap *regmap; + struct snd_soc_jack *jack; + struct input_dev *button_dev; + bool plugged; + bool mic_present; +}; + +/* TS3A227E registers */ +#define TS3A227E_REG_DEVICE_ID 0x00 +#define TS3A227E_REG_INTERRUPT 0x01 +#define TS3A227E_REG_KP_INTERRUPT 0x02 +#define TS3A227E_REG_INTERRUPT_DISABLE 0x03 +#define TS3A227E_REG_SETTING_1 0x04 +#define TS3A227E_REG_SETTING_2 0x05 +#define TS3A227E_REG_SETTING_3 0x06 +#define TS3A227E_REG_SWITCH_CONTROL_1 0x07 +#define TS3A227E_REG_SWITCH_CONTROL_2 0x08 +#define TS3A227E_REG_SWITCH_STATUS_1 0x09 +#define TS3A227E_REG_SWITCH_STATUS_2 0x0a +#define TS3A227E_REG_ACCESSORY_STATUS 0x0b +#define TS3A227E_REG_ADC_OUTPUT 0x0c +#define TS3A227E_REG_KP_THRESHOLD_1 0x0d +#define TS3A227E_REG_KP_THRESHOLD_2 0x0e +#define TS3A227E_REG_KP_THRESHOLD_3 0x0f + +/* TS3A227E_REG_INTERRUPT 0x01 */ +#define INS_REM_EVENT 0x01 +#define DETECTION_COMPLETE_EVENT 0x02 + +/* TS3A227E_REG_KP_INTERRUPT 0x02 */ +#define PRESS_MASK(idx) (0x01 << (2 * (idx))) +#define RELEASE_MASK(idx) (0x02 << (2 * (idx))) + +/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */ +#define INS_REM_INT_DISABLE 0x01 +#define DETECTION_COMPLETE_INT_DISABLE 0x02 +#define ADC_COMPLETE_INT_DISABLE 0x04 +#define INTB_DISABLE 0x08 + +/* TS3A227E_REG_SETTING_2 0x05 */ +#define KP_ENABLE 0x04 + +/* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ +#define TYPE_3_POLE 0x01 +#define TYPE_4_POLE_OMTP 0x02 +#define TYPE_4_POLE_STANDARD 0x04 +#define JACK_INSERTED 0x08 +#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD) + +/* Button values to be reported on the jack */ +static const int ts3a227e_keycodes[] = { + KEY_MEDIA, + KEY_VOLUMEUP, + KEY_VOLUMEDOWN, + KEY_VOICECOMMAND, +}; + +#define TS3A227E_NUM_KEYS ARRAY_SIZE(ts3a227e_keycodes) +#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | SND_JACK_MICROPHONE) + +static const struct reg_default ts3a227e_reg_defaults[] = { + { TS3A227E_REG_DEVICE_ID, 0x10 }, + { TS3A227E_REG_INTERRUPT, 0x00 }, + { TS3A227E_REG_KP_INTERRUPT, 0x00 }, + { TS3A227E_REG_INTERRUPT_DISABLE, 0x08 }, + { TS3A227E_REG_SETTING_1, 0x23 }, + { TS3A227E_REG_SETTING_2, 0x00 }, + { TS3A227E_REG_SETTING_3, 0x0e }, + { TS3A227E_REG_SWITCH_CONTROL_1, 0x00 }, + { TS3A227E_REG_SWITCH_CONTROL_2, 0x00 }, + { TS3A227E_REG_SWITCH_STATUS_1, 0x0c }, + { TS3A227E_REG_SWITCH_STATUS_2, 0x00 }, + { TS3A227E_REG_ACCESSORY_STATUS, 0x00 }, + { TS3A227E_REG_ADC_OUTPUT, 0x00 }, + { TS3A227E_REG_KP_THRESHOLD_1, 0x20 }, + { TS3A227E_REG_KP_THRESHOLD_2, 0x40 }, + { TS3A227E_REG_KP_THRESHOLD_3, 0x68 }, +}; + +static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2: + case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3: + return true; + default: + return false; + } +} + +static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: + case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: + return true; + default: + return false; + } +} + +static void ts3a227e_jack_report(struct ts3a227e *ts3a227e) +{ + int report = 0; + + if (!ts3a227e->jack) + return; + + if (ts3a227e->plugged) + report = SND_JACK_HEADPHONE; + if (ts3a227e->mic_present) + report |= SND_JACK_MICROPHONE; + snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK); +} + +static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg) +{ + bool plugged, mic_present; + unsigned int i; + + plugged = !!(acc_reg & JACK_INSERTED); + mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK); + + ts3a227e->plugged = plugged; + + if (mic_present != ts3a227e->mic_present) { + ts3a227e->mic_present = mic_present; + for (i = 0; i < TS3A227E_NUM_KEYS; i++) + input_report_key(ts3a227e->button_dev, + ts3a227e_keycodes[i], 0); + /* Enable key press detection. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_2, + KP_ENABLE, mic_present ? KP_ENABLE : 0); + } +} + +static irqreturn_t ts3a227e_interrupt(int irq, void *data) +{ + struct ts3a227e *ts3a227e = (struct ts3a227e *)data; + struct regmap *regmap = ts3a227e->regmap; + unsigned int int_reg, kp_int_reg, acc_reg, i; + + /* Check for plug/unplug. */ + regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg); + if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) { + regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg); + ts3a227e_new_jack_state(ts3a227e, acc_reg); + } + + /* Report any key events. */ + regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg); + for (i = 0; i < TS3A227E_NUM_KEYS; i++) { + if (kp_int_reg & PRESS_MASK(i)) + input_report_key(ts3a227e->button_dev, + ts3a227e_keycodes[i], 1); + if (kp_int_reg & RELEASE_MASK(i)) + input_report_key(ts3a227e->button_dev, + ts3a227e_keycodes[i], 0); + } + + input_sync(ts3a227e->button_dev); + ts3a227e_jack_report(ts3a227e); + + return IRQ_HANDLED; +} + +/** + * ts3a227e_enable_jack_detect - Specify a jack for event reporting + * + * @codec: codec to register the jack with + * @jack: jack to use to report headphone/mic plug/unplug on + * + * After this function has been called the headset insert/remove will be routed + * to the given jack. Jack can be null to stop reporting. + */ +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct ts3a227e *ts3a227e = snd_soc_codec_get_drvdata(codec); + + ts3a227e->jack = jack; + ts3a227e_jack_report(ts3a227e); + + return 0; +} +EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); + +static struct snd_soc_codec_driver ts3a227e_codec_driver; + +static const struct regmap_config ts3a227e_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = TS3A227E_REG_KP_THRESHOLD_3, + .readable_reg = ts3a227e_readable_reg, + .writeable_reg = ts3a227e_writeable_reg, + .volatile_reg = ts3a227e_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ts3a227e_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), +}; + +static int ts3a227e_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ts3a227e *ts3a227e; + struct device *dev = &i2c->dev; + unsigned int i; + int ret; + + ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL); + if (!ts3a227e) + return -ENOMEM; + + i2c_set_clientdata(i2c, ts3a227e); + + ts3a227e->button_dev = devm_input_allocate_device(dev); + if (!ts3a227e->button_dev) { + dev_err(dev, "Failed to allocate button_dev\n"); + return -ENOMEM; + } + + ts3a227e->button_dev->evbit[0] = BIT_MASK(EV_KEY); + for (i = 0; i < TS3A227E_NUM_KEYS; i++) + ts3a227e->button_dev->keybit[BIT_WORD(ts3a227e_keycodes[i])] |= + BIT_MASK(ts3a227e_keycodes[i]); + + ts3a227e->button_dev->name = "ts3a227e headset buttons"; + ret = input_register_device(ts3a227e->button_dev); + if (ret) { + dev_err(dev, "Failed to register input dev %d\n", ret); + return ret; + } + + ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config); + if (IS_ERR(ts3a227e->regmap)) + return PTR_ERR(ts3a227e->regmap); + + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "TS3A227E", ts3a227e); + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &ts3a227e_codec_driver, + NULL, 0); + if (ret) + return ret; + + /* Enable interrupts except for ADC complete. */ + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE, + INTB_DISABLE | ADC_COMPLETE_INT_DISABLE, + ADC_COMPLETE_INT_DISABLE); + + return 0; +} + +static int ts3a227e_i2c_remove(struct i2c_client *i2c) +{ + struct ts3a227e *ts3a227e = i2c_get_clientdata(i2c); + + input_unregister_device(ts3a227e->button_dev); + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id ts3a227e_i2c_ids[] = { + { "ts3a227e", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids); + +static const struct of_device_id ts3a227e_of_match[] = { + { .compatible = "ti,ts3a227e", }, + { } +}; +MODULE_DEVICE_TABLE(of, ts3a227e_of_match); + +static struct i2c_driver ts3a227e_driver = { + .driver = { + .name = "ts3a227e", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ts3a227e_of_match), + }, + .probe = ts3a227e_i2c_probe, + .remove = ts3a227e_i2c_remove, + .id_table = ts3a227e_i2c_ids, +}; +module_i2c_driver(ts3a227e_driver); + +MODULE_DESCRIPTION("ASoC ts3a227e driver"); +MODULE_AUTHOR("Dylan Reid <dgreid(a)chromium.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h new file mode 100644 index 0000000..8a58886 --- /dev/null +++ b/sound/soc/codecs/ts3a227e.h @@ -0,0 +1,17 @@ +/* + * TS3A227E Autonous Audio Accessory Detection and Configureation Switch + * + * Copyright (C) 2014 Google, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TS3A227E_H +#define _TS3A227E_H + +int ts3a227e_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + +#endif -- 2.1.2.330.g565301e

2 2

[alsa-devel] [PATCH] ASoC: arizona: Add DSP_B and LEFT_J mode support
by Charles Keepax 12 Nov '14

12 Nov '14

From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 28 +++++++++++++++++++++++++--- 1 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 1e352d8..9550d74 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -61,6 +61,11 @@ #define ARIZONA_FLL_MIN_OUTDIV 2 #define ARIZONA_FLL_MAX_OUTDIV 7 +#define ARIZONA_FMT_DSP_MODE_A 0 +#define ARIZONA_FMT_DSP_MODE_B 1 +#define ARIZONA_FMT_I2S_MODE 2 +#define ARIZONA_FMT_LEFT_JUSTIFIED_MODE 3 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -946,10 +951,26 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - mode = 0; + mode = ARIZONA_FMT_DSP_MODE_A; + break; + case SND_SOC_DAIFMT_DSP_B: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "DSP_B not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_DSP_MODE_B; break; case SND_SOC_DAIFMT_I2S: - mode = 2; + mode = ARIZONA_FMT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "LEFT_J not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_LEFT_JUSTIFIED_MODE; break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", @@ -1298,7 +1319,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { + val &= ARIZONA_AIF1_FMT_MASK; + if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); bclk_target /= channels; bclk_target *= channels + 1; -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: add missing description for kerneldoc
by Takashi Sakamoto 12 Nov '14

12 Nov '14

A description for second paramter for cmp_connection_check_used() is missing. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/cmp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index ba8df5a..dcccb01 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -154,6 +154,7 @@ EXPORT_SYMBOL(cmp_connection_init); /** * cmp_connection_check_used - check connection is already esablished or not * @c: the connection manager to be checked + * @used: the pointer to store result to check the connection */ int cmp_connection_check_used(struct cmp_connection *c, bool *used) { -- 2.1.0

2 2

[alsa-devel] [PATCHv2] ASoC: cs42l51: re-hook of_match_table pointer
by Thomas Petazzoni 12 Nov '14

12 Nov '14

In commit a1253ef6d3fa ("ASoC: cs42l51: split i2c from codec driver"), the I2C part of the CS42L51 was moved to a separate file, but the definition of the of_device_id array was left in the driver file itself, no longer connected to the platform_driver structure using the .of_match_table pointer. This commit exports the of_device_id array in cs42l51, and uses it as .of_match_able in cs42l51-i2c.c. This solution was suggested by Brian Austin. Fixes: a1253ef6d3fa ("ASoC: cs42l51: split i2c from codec driver") Cc: <stable(a)vger.kernel.org> # v3.16+ Cc: Arnaud Patard (Rtp) <arnaud.patard(a)rtp-net.org> Cc: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> --- sound/soc/codecs/cs42l51-i2c.c | 1 + sound/soc/codecs/cs42l51.c | 4 +++- sound/soc/codecs/cs42l51.h | 1 + 3 files changed, 5 insertions(+), 1 deletion(-) --- Changes since v1: - Instead of moving the of_device_id array definition to the I2C specific file, keep it in the core file, and export it so it can be used by the I2C specific file. Suggested by Brian Austin. diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index cee51ae..c40428f 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -46,6 +46,7 @@ static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51", .owner = THIS_MODULE, + .of_match_table = cs42l51_of_match, }, .probe = cs42l51_i2c_probe, .remove = cs42l51_i2c_remove, diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 09488d9..669c38f 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -558,11 +558,13 @@ error: } EXPORT_SYMBOL_GPL(cs42l51_probe); -static const struct of_device_id cs42l51_of_match[] = { +const struct of_device_id cs42l51_of_match[] = { { .compatible = "cirrus,cs42l51", }, { } }; MODULE_DEVICE_TABLE(of, cs42l51_of_match); +EXPORT_SYMBOL_GPL(cs42l51_of_match); + MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 8c55bf3..0ca8054 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -22,6 +22,7 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); +extern const struct of_device_id cs42l51_of_match[]; #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 -- 2.0.0

3 2

[alsa-devel] [PATCH] ASoC: arizona: Add support for 768kHz DMIC operation
by Charles Keepax 12 Nov '14

12 Nov '14

The new IPs supports a new lower frequency 768kHz DMIC operation add support for this into the OSR control. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 730636c..1e352d8 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -648,7 +648,7 @@ SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum, EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); static const char * const arizona_in_dmic_osr_text[] = { - "1.536MHz", "3.072MHz", "6.144MHz", + "1.536MHz", "3.072MHz", "6.144MHz", "768kHz", }; const struct soc_enum arizona_in_dmic_osr[] = { -- 1.7.2.5

2 1

[alsa-devel] [PATCH] a52: Ignore start request if we're already running
by David Henningsson 12 Nov '14

12 Nov '14

When trying to start a PCM that's already running, the result is EBADFD. We might have implicitly started the buffer by filling it up, so just ignore this request if we're already running. Reported-by: Chow Loong Jin <hyperair(a)ubuntu.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- a52/pcm_a52.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/a52/pcm_a52.c b/a52/pcm_a52.c index b467ec8..59e25cb 100644 --- a/a52/pcm_a52.c +++ b/a52/pcm_a52.c @@ -482,6 +482,12 @@ static int a52_start(snd_pcm_ioplug_t *io) { struct a52_ctx *rec = io->private_data; + /* When trying to start a PCM that's already running, the result is + EBADFD. We might have implicitly started the buffer by filling it + up, so just ignore this request if we're already running. */ + if (snd_pcm_state(rec->slave) == SND_PCM_STATE_RUNNING) + return 0; + return snd_pcm_start(rec->slave); } -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: Add overrun/underrun event handling
by Peter Ujfalusi 12 Nov '14

12 Nov '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> An underrun (playback) event occurs when the serializer transfer data from the XRBUF buffer to the XRSR shift register, but the XRBUF hasn't been filled. Similarly, the overrun (capture) event occurs when data from the XRSR shift register is transferred to the XRBUF but it hasn't been read yet. These events are handled as XRUN events that cause the pcm to stop. The stream has to be explicitly restarted by the userspace which ensures that after stopping/starting McASP the data transfer is aligned with DMA. The other possibility was to internally stop and start McASP without DMA even knowing about it. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Chanhes since v2: - From interrupt handler return IRQ_HANDLED or IRQ_NONE depending on if we have handled event or not. Changes since v1: - only ack handled interrupt (over/underflow at the moment) - Use variable to keep track of requested irq sources for future use - print warning in case of not handled interrupt. Regards, Peter .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 124 +++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 11 ++ 3 files changed, 136 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 60ca07996458..46bc9829c71a 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -32,7 +32,7 @@ Optional properties: - rx-num-evt : FIFO levels. - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP, currently not used by the driver +- interrupts : Interrupt numbers for McASP - interrupt-names : Known interrupt names are "tx" and "rx" - pinctrl-0: Should specify pin control group used for this controller. - pinctrl-names: Should contain only one value - "default", for more details diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6b15a974f585..9de954a24160 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -70,6 +70,7 @@ struct davinci_mcasp { void __iomem *base; u32 fifo_base; struct device *dev; + struct snd_pcm_substream *substreams[2]; /* McASP specific data */ int tdm_slots; @@ -80,6 +81,7 @@ struct davinci_mcasp { u8 bclk_div; u16 bclk_lrclk_ratio; int streams; + u32 irq_request[2]; int sysclk_freq; bool bclk_master; @@ -185,6 +187,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); if (mcasp_is_synchronous(mcasp)) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable receive IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -214,6 +220,10 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); /* Release Frame Sync generator */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable transmit IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) @@ -228,6 +238,10 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); + /* * In synchronous mode stop the TX clocks if no other stream is * running @@ -249,6 +263,10 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { u32 val = 0; + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); + /* * In synchronous mode keep TX clocks running if the capture stream is * still running. @@ -276,6 +294,76 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) mcasp_stop_rx(mcasp); } +static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG); + if (stat & XUNDRN & irq_mask) { + dev_warn(mcasp->dev, "Transmit buffer underflow\n"); + handled_mask |= XUNDRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask); + + return IRQ_RETVAL(handled_mask); +} + +static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG); + if (stat & ROVRN & irq_mask) { + dev_warn(mcasp->dev, "Receive buffer overflow\n"); + handled_mask |= ROVRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask); + + return IRQ_RETVAL(handled_mask); +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -878,6 +966,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir; + mcasp->substreams[substream->stream] = substream; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; @@ -916,6 +1006,8 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + mcasp->substreams[substream->stream] = NULL; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return; @@ -1266,6 +1358,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; + char *irq_name; + int irq; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -1346,6 +1440,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "rx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_rx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "RX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + + irq = platform_get_irq_byname(pdev, "tx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_tx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "TX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + } + dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); if (dat) mcasp->dat_port = true; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9737108f0305..79dc511180bf 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -256,6 +256,7 @@ * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits */ +#define XRERR BIT(8) /* Transmit/Receive error */ #define XRDATA BIT(5) /* Transmit/Receive data ready */ /* @@ -285,6 +286,16 @@ #define TXDATADMADIS BIT(0) /* + * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits + */ +#define ROVRN BIT(0) + +/* + * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits + */ +#define XUNDRN BIT(0) + +/* * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits */ #define FIFO_ENABLE BIT(16) -- 2.1.3

2 1

[alsa-devel] [PATCH v7 00/20] Rework OMAP4+ HDMI audio support
by Jyri Sarha 12 Nov '14

12 Nov '14

The patches are based on: git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git for-next The base, the patches, and couple of additional not-to-be-merged omap2plus_defconfig patches can be found here: https://github.com/jsarha/linux.git omap-hdmi-audio It would make the most sense to get these in trough fbdev tree. So it would be nice to get acked-bys (if the patches are Ok) for ASoC side changes from appropriate maintainers. Changes since v6: - "ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support" - Fix Kconfig help text - Remove #include <sound/simple_card.h> - Use snd-soc-dummy codec instead of hdmi-audio-codec - Add: OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() - Add: OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter - Add: "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - "OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - "OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - Add: "OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback" Jyri Sarha (20): OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi: Remove most of OMAP[45]_DSS_HDMI_AUDIO ifdefs OMAPDSS: hdmi4_core: Remove unused hdmi4_audio_get_dma_port() OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device OMAPDSS: hdmi_wp: Add function for getting audio dma address OMAPDSS: hdmi: Make hdmi structure public OMAPDSS: hdmi: Add pdev pointer for audio_pdev in HDMI DRV data ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support OMAPDSS: Kconfig: Remove HDMI audio booleans from Kconfig OMAPDSS: hdmi: Make hdmi_mode_has_audio() more user friedly OMAPDSS: hdmi.h: Add members to hdmi drvdata for audio implementation OMAPDSS: hdmi4: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove all references to obsolete HDMI audio callbacks OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ----- .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 7 - drivers/video/fbdev/omap2/dss/hdmi.h | 38 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 269 +++++++------ drivers/video/fbdev/omap2/dss/hdmi4_core.c | 14 - drivers/video/fbdev/omap2/dss/hdmi4_core.h | 4 - drivers/video/fbdev/omap2/dss/hdmi5.c | 265 ++++++------- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 9 +- drivers/video/fbdev/omap2/dss/hdmi5_core.h | 2 - drivers/video/fbdev/omap2/dss/hdmi_common.c | 2 - drivers/video/fbdev/omap2/dss/hdmi_wp.c | 16 +- include/sound/omap-hdmi-audio.h | 43 +++ include/video/omapdss.h | 40 -- sound/soc/omap/Kconfig | 26 +- sound/soc/omap/Makefile | 6 +- sound/soc/omap/omap-hdmi-audio.c | 407 ++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 ----------------- sound/soc/omap/omap-hdmi.h | 38 -- 20 files changed, 781 insertions(+), 1011 deletions(-) create mode 100644 include/sound/omap-hdmi-audio.h create mode 100644 sound/soc/omap/omap-hdmi-audio.c delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h -- 1.7.9.5

2 23

[alsa-devel] [PATCH 4/4] ASoC: rt5670: add codec irq support
by bardliao＠realtek.com 12 Nov '14

12 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch provide the jack function of rt5670. We use a irq pin to notice the jack event. rt5670_irq_detection will determine which status should be reported. We can get gpio information from gpio number or gpiod_dev. That's why we define both on platform data. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5670.h | 2 + sound/soc/codecs/rt5670.c | 176 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5670.h | 2 + 3 files changed, 178 insertions(+), 2 deletions(-) diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h index bd31119..bb5cf6d 100644 --- a/include/sound/rt5670.h +++ b/include/sound/rt5670.h @@ -14,6 +14,8 @@ struct rt5670_platform_data { int jd_mode; bool in2_diff; + int codec_gpio; + bool dev_gpio; bool dmic_en; unsigned int dmic1_data_pin; diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 89fa507..cb5a91c 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -401,6 +401,144 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg) } } +/** + * rt5670_headset_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert) +{ + int val; + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD, + RT5670_CBJ_MN_JD); + snd_soc_write(codec, RT5670_GPIO_CTRL2, 0x0004); + snd_soc_update_bits(codec, RT5670_GPIO_CTRL1, + RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); + snd_soc_update_bits(codec, RT5670_CJ_CTRL1, + RT5670_CBJ_BST1_EN, RT5670_CBJ_BST1_EN); + snd_soc_write(codec, RT5670_JD_CTRL3, 0x00f0); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, RT5670_CBJ_MN_JD); + snd_soc_update_bits(codec, RT5670_CJ_CTRL2, + RT5670_CBJ_MN_JD, 0); + msleep(300); + val = snd_soc_read(codec, RT5670_CJ_CTRL3) & 0x7; + if (val == 0x1 || val == 0x2) { + rt5670->jack_type = SND_JACK_HEADSET; + /* for push button */ + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5670_IL_CMD, 0x40, 0x40); + snd_soc_read(codec, RT5670_IL_CMD); + } else { + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = SND_JACK_HEADPHONE; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + } else { + snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0); + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4); + rt5670->jack_type = 0; + snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); + snd_soc_dapm_sync(&codec->dapm); + } + + return rt5670->jack_type; +} + +static int rt5670_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5670_IL_CMD); + btn_type = val & 0xff80; + snd_soc_write(codec, RT5670_IL_CMD, val); + if (btn_type != 0) { + msleep(20); + val = snd_soc_read(codec, RT5670_IL_CMD); + snd_soc_write(codec, RT5670_IL_CMD, val); + } + + return btn_type; +} + +static int rt5670_irq_detection(void *data) +{ + struct rt5670_priv *rt5670 = (struct rt5670_priv *)data; + struct snd_soc_jack_gpio *gpio = &rt5670->hp_gpio; + struct snd_soc_jack *jack = &rt5670->jack; + int val, btn_type, report = jack->status; + + if (rt5670->pdata.jd_mode == 1) /* 2 port */ + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0070; + else + val = snd_soc_read(rt5670->codec, RT5670_A_JD_CTRL1) & 0x0020; + + switch (val) { + /* jack in */ + case 0x30: /* 2 port */ + case 0x0: /* 1 port or 2 port */ + if (rt5670->jack_type == 0) { + report = rt5670_headset_detect(rt5670->codec, 1); + /* for push button and jack out */ + gpio->debounce_time = 25; + break; + } + btn_type = 0; + if (snd_soc_read(rt5670->codec, RT5670_INT_IRQ_ST) & 0x4) { + /* button pressed */ + report = SND_JACK_HEADSET; + btn_type = rt5670_button_detect(rt5670->codec); + switch (btn_type) { + case 0x2000: /* up */ + report |= SND_JACK_BTN_1; + break; + case 0x0400: /* center */ + report |= SND_JACK_BTN_0; + break; + case 0x0080: /* down */ + report |= SND_JACK_BTN_2; + break; + default: + dev_err(rt5670->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + if (btn_type == 0)/* button release */ + report = rt5670->jack_type; + + break; + /* jack out */ + case 0x70: /* 2 port */ + case 0x10: /* 2 port */ + case 0x20: /* 1 port */ + report = 0; + snd_soc_update_bits(rt5670->codec, RT5670_INT_IRQ_ST, 0x1, 0x0); + rt5670_headset_detect(rt5670->codec, 0); + gpio->debounce_time = 150; /* for jack in */ + break; + default: + break; + } + + return report; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); @@ -2413,6 +2551,34 @@ static int rt5670_probe(struct snd_soc_codec *codec) return -ENODEV; } rt5670->codec = codec; + if ((rt5670->pdata.codec_gpio != -1) || (rt5670->pdata.dev_gpio)) { + if (rt5670->pdata.dev_gpio) + rt5670->hp_gpio.gpiod_dev = codec->dev; + else + rt5670->hp_gpio.gpio = rt5670->pdata.codec_gpio; + + rt5670->hp_gpio.name = "headphone detect"; + rt5670->hp_gpio.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2; + rt5670->hp_gpio.debounce_time = 150, + rt5670->hp_gpio.wake = true, + rt5670->hp_gpio.data = (struct rt5670_priv *)rt5670, + rt5670->hp_gpio.jack_status_check = rt5670_irq_detection, + ret = snd_soc_jack_new(codec, rt5670->hp_gpio.name, + rt5670->hp_gpio.report, + &rt5670->jack); + if (ret) { + dev_err(codec->dev, "Jack creation failed\n"); + return ret; + } + + ret = snd_soc_jack_add_gpios(&rt5670->jack, 1, + &rt5670->hp_gpio); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + } return 0; } @@ -2422,6 +2588,7 @@ static int rt5670_remove(struct snd_soc_codec *codec) struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); regmap_write(rt5670->regmap, RT5670_RESET, 0); + snd_soc_jack_free_gpios(&rt5670->jack, 1, &rt5670->hp_gpio); return 0; } @@ -2590,12 +2757,17 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5670->regmap, RT5670_IN2, RT5670_IN_DF2, RT5670_IN_DF2); - if (i2c->irq) { + if ((rt5670->pdata.codec_gpio != -1) || (rt5670->pdata.dev_gpio)) { + /* for push button */ + regmap_write(rt5670->regmap, RT5670_IL_CMD, 0x0000); + regmap_write(rt5670->regmap, RT5670_IL_CMD2, 0x0010); + regmap_write(rt5670->regmap, RT5670_IL_CMD3, 0x0014); + /* for irq */ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1, RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_IRQ); regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2, RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT); - + regmap_update_bits(rt5670->regmap, RT5670_DIG_MISC, 0x8, 0x8); } if (rt5670->pdata.jd_mode) { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index d11b9c2..54933f3 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1987,6 +1987,8 @@ struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; struct regmap *regmap; + struct snd_soc_jack jack; + struct snd_soc_jack_gpio hp_gpio; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

3 2

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Add overrun/underrun event handling
by Peter Ujfalusi 12 Nov '14

12 Nov '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> An underrun (playback) event occurs when the serializer transfer data from the XRBUF buffer to the XRSR shift register, but the XRBUF hasn't been filled. Similarly, the overrun (capture) event occurs when data from the XRSR shift register is transferred to the XRBUF but it hasn't been read yet. These events are handled as XRUN events that cause the pcm to stop. The stream has to be explicitly restarted by the userspace which ensures that after stopping/starting McASP the data transfer is aligned with DMA. The other possibility was to internally stop and start McASP without DMA even knowing about it. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - only ack handled interrupt (over/underflow at the moment) - Use variable to keep track of requested irq sources for future use - print warning in case of not handled interrupt. Regards, Peter .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 124 +++++++++++++++++++++ sound/soc/davinci/davinci-mcasp.h | 11 ++ 3 files changed, 136 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 60ca07996458..46bc9829c71a 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -32,7 +32,7 @@ Optional properties: - rx-num-evt : FIFO levels. - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP, currently not used by the driver +- interrupts : Interrupt numbers for McASP - interrupt-names : Known interrupt names are "tx" and "rx" - pinctrl-0: Should specify pin control group used for this controller. - pinctrl-names: Should contain only one value - "default", for more details diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6b15a974f585..601268350c58 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -70,6 +70,7 @@ struct davinci_mcasp { void __iomem *base; u32 fifo_base; struct device *dev; + struct snd_pcm_substream *substreams[2]; /* McASP specific data */ int tdm_slots; @@ -80,6 +81,7 @@ struct davinci_mcasp { u8 bclk_div; u16 bclk_lrclk_ratio; int streams; + u32 irq_request[2]; int sysclk_freq; bool bclk_master; @@ -185,6 +187,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); if (mcasp_is_synchronous(mcasp)) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable receive IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); } static void mcasp_start_tx(struct davinci_mcasp *mcasp) @@ -214,6 +220,10 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); /* Release Frame Sync generator */ mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + + /* enable transmit IRQs */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); } static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) @@ -228,6 +238,10 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream) static void mcasp_stop_rx(struct davinci_mcasp *mcasp) { + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]); + /* * In synchronous mode stop the TX clocks if no other stream is * running @@ -249,6 +263,10 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp) { u32 val = 0; + /* disable IRQ sources */ + mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG, + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]); + /* * In synchronous mode keep TX clocks running if the capture stream is * still running. @@ -276,6 +294,76 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream) mcasp_stop_rx(mcasp); } +static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG); + if (stat & XUNDRN & irq_mask) { + dev_warn(mcasp->dev, "Transmit buffer underflow\n"); + handled_mask |= XUNDRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask); + + return IRQ_HANDLED; +} + +static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + struct snd_pcm_substream *substream; + u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]; + u32 handled_mask = 0; + u32 stat; + + stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG); + if (stat & ROVRN & irq_mask) { + dev_warn(mcasp->dev, "Receive buffer overflow\n"); + handled_mask |= ROVRN; + + substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]; + if (substream) { + snd_pcm_stream_lock_irq(substream); + if (snd_pcm_running(substream)) + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irq(substream); + } + } + + if (!handled_mask) + dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n", + stat); + + if (stat & XRERR) + handled_mask |= XRERR; + + /* Ack the handled event only */ + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask); + + return IRQ_HANDLED; +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -878,6 +966,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir; + mcasp->substreams[substream->stream] = substream; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; @@ -916,6 +1006,8 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + mcasp->substreams[substream->stream] = NULL; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return; @@ -1266,6 +1358,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; + char *irq_name; + int irq; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -1346,6 +1440,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "rx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_rx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "RX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + + irq = platform_get_irq_byname(pdev, "tx"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_tx_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "TX IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + } + dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); if (dat) mcasp->dat_port = true; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9737108f0305..79dc511180bf 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -256,6 +256,7 @@ * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits */ +#define XRERR BIT(8) /* Transmit/Receive error */ #define XRDATA BIT(5) /* Transmit/Receive data ready */ /* @@ -285,6 +286,16 @@ #define TXDATADMADIS BIT(0) /* + * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits + */ +#define ROVRN BIT(0) + +/* + * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits + */ +#define XUNDRN BIT(0) + +/* * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits */ #define FIFO_ENABLE BIT(16) -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Add TDM channel mux in DAC side of IF1 and IF2
by Oder Chiou 12 Nov '14

12 Nov '14

It is the slot selection in DAC side of IF1 and IF2. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 330 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5677.h | 48 ++++++- 2 files changed, 358 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 4b6f7d5..5d317c68 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1906,6 +1906,126 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if2_adc_tdm_swap_mux = SOC_DAPM_ENUM("IF2 ADC TDM Swap Source", rt5677_if2_adc_tdm_swap_enum); +/* TDM IF1/2 DAC Data Selection */ /* MX-3E[14:12][10:8][6:4][2:0] + MX-3F[14:12][10:8][6:4][2:0] + MX-43[14:12][10:8][6:4][2:0] + MX-44[14:12][10:8][6:4][2:0] */ +static const char * const rt5677_if12_dac_tdm_sel_src[] = { + "Slot0", "Slot1", "Slot2", "Slot3", "Slot4", "Slot5", "Slot6", "Slot7" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac0_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC0_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac0_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC0 TDM Source", rt5677_if1_dac0_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac1_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC1_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac1_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC1 TDM Source", rt5677_if1_dac1_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac2_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC2_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac2_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC2 TDM Source", rt5677_if1_dac2_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac3_tdm_sel_enum, RT5677_TDM1_CTRL4, + RT5677_IF1_DAC3_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac3_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC3 TDM Source", rt5677_if1_dac3_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac4_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC4_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac4_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC4 TDM Source", rt5677_if1_dac4_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac5_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC5_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac5_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC5 TDM Source", rt5677_if1_dac5_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac6_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC6_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac6_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC6 TDM Source", rt5677_if1_dac6_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if1_dac7_tdm_sel_enum, RT5677_TDM1_CTRL5, + RT5677_IF1_DAC7_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if1_dac7_tdm_sel_mux = + SOC_DAPM_ENUM("IF1 DAC7 TDM Source", rt5677_if1_dac7_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac0_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC0_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac0_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC0 TDM Source", rt5677_if2_dac0_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac1_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC1_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac1_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC1 TDM Source", rt5677_if2_dac1_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac2_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC2_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac2_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC2 TDM Source", rt5677_if2_dac2_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac3_tdm_sel_enum, RT5677_TDM2_CTRL4, + RT5677_IF2_DAC3_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac3_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC3 TDM Source", rt5677_if2_dac3_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac4_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC4_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac4_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC4 TDM Source", rt5677_if2_dac4_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac5_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC5_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac5_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC5 TDM Source", rt5677_if2_dac5_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac6_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC6_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac6_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC6 TDM Source", rt5677_if2_dac6_tdm_sel_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5677_if2_dac7_tdm_sel_enum, RT5677_TDM2_CTRL5, + RT5677_IF2_DAC7_SFT, rt5677_if12_dac_tdm_sel_src); + +static const struct snd_kcontrol_new rt5677_if2_dac7_tdm_sel_mux = + SOC_DAPM_ENUM("IF2 DAC7 TDM Source", rt5677_if2_dac7_tdm_sel_enum); + static int rt5677_bst1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2389,6 +2509,40 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { SND_SOC_DAPM_MUX("SLB ADC4 Mux", SND_SOC_NOPM, 0, 0, &rt5677_slb_adc4_mux), + SND_SOC_DAPM_MUX("IF1 DAC0 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac4_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC5 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac5_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC6 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac6_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF1 DAC7 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if1_dac7_tdm_sel_mux), + + SND_SOC_DAPM_MUX("IF2 DAC0 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC1 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC2 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC3 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC4 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac4_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC5 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac5_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC6 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac6_tdm_sel_mux), + SND_SOC_DAPM_MUX("IF2 DAC7 Mux", SND_SOC_NOPM, 0, 0, + &rt5677_if2_dac7_tdm_sel_mux), + /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), @@ -3036,14 +3190,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF1 DAC6", NULL, "I2S1" }, { "IF1 DAC7", NULL, "I2S1" }, - { "IF1 DAC01", NULL, "IF1 DAC0" }, - { "IF1 DAC01", NULL, "IF1 DAC1" }, - { "IF1 DAC23", NULL, "IF1 DAC2" }, - { "IF1 DAC23", NULL, "IF1 DAC3" }, - { "IF1 DAC45", NULL, "IF1 DAC4" }, - { "IF1 DAC45", NULL, "IF1 DAC5" }, - { "IF1 DAC67", NULL, "IF1 DAC6" }, - { "IF1 DAC67", NULL, "IF1 DAC7" }, + { "IF1 DAC0 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC0 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC0 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC0 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC0 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC0 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC0 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC0 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC1 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC1 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC1 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC1 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC1 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC1 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC1 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC1 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC2 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC2 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC2 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC2 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC2 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC2 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC2 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC2 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC3 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC3 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC3 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC3 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC3 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC3 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC3 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC3 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC4 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC4 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC4 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC4 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC4 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC4 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC4 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC4 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC5 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC5 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC5 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC5 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC5 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC5 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC5 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC5 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC6 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC6 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC6 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC6 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC6 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC6 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC6 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC6 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC7 Mux", "Slot0", "IF1 DAC0" }, + { "IF1 DAC7 Mux", "Slot1", "IF1 DAC1" }, + { "IF1 DAC7 Mux", "Slot2", "IF1 DAC2" }, + { "IF1 DAC7 Mux", "Slot3", "IF1 DAC3" }, + { "IF1 DAC7 Mux", "Slot4", "IF1 DAC4" }, + { "IF1 DAC7 Mux", "Slot5", "IF1 DAC5" }, + { "IF1 DAC7 Mux", "Slot6", "IF1 DAC6" }, + { "IF1 DAC7 Mux", "Slot7", "IF1 DAC7" }, + + { "IF1 DAC01", NULL, "IF1 DAC0 Mux" }, + { "IF1 DAC01", NULL, "IF1 DAC1 Mux" }, + { "IF1 DAC23", NULL, "IF1 DAC2 Mux" }, + { "IF1 DAC23", NULL, "IF1 DAC3 Mux" }, + { "IF1 DAC45", NULL, "IF1 DAC4 Mux" }, + { "IF1 DAC45", NULL, "IF1 DAC5 Mux" }, + { "IF1 DAC67", NULL, "IF1 DAC6 Mux" }, + { "IF1 DAC67", NULL, "IF1 DAC7 Mux" }, { "IF2 DAC0", NULL, "AIF2RX" }, { "IF2 DAC1", NULL, "AIF2RX" }, @@ -3062,14 +3288,86 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "IF2 DAC6", NULL, "I2S2" }, { "IF2 DAC7", NULL, "I2S2" }, - { "IF2 DAC01", NULL, "IF2 DAC0" }, - { "IF2 DAC01", NULL, "IF2 DAC1" }, - { "IF2 DAC23", NULL, "IF2 DAC2" }, - { "IF2 DAC23", NULL, "IF2 DAC3" }, - { "IF2 DAC45", NULL, "IF2 DAC4" }, - { "IF2 DAC45", NULL, "IF2 DAC5" }, - { "IF2 DAC67", NULL, "IF2 DAC6" }, - { "IF2 DAC67", NULL, "IF2 DAC7" }, + { "IF2 DAC0 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC0 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC0 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC0 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC0 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC0 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC0 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC0 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC1 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC1 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC1 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC1 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC1 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC1 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC1 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC1 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC2 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC2 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC2 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC2 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC2 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC2 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC2 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC2 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC3 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC3 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC3 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC3 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC3 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC3 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC3 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC3 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC4 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC4 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC4 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC4 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC4 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC4 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC4 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC4 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC5 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC5 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC5 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC5 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC5 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC5 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC5 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC5 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC6 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC6 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC6 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC6 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC6 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC6 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC6 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC6 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC7 Mux", "Slot0", "IF2 DAC0" }, + { "IF2 DAC7 Mux", "Slot1", "IF2 DAC1" }, + { "IF2 DAC7 Mux", "Slot2", "IF2 DAC2" }, + { "IF2 DAC7 Mux", "Slot3", "IF2 DAC3" }, + { "IF2 DAC7 Mux", "Slot4", "IF2 DAC4" }, + { "IF2 DAC7 Mux", "Slot5", "IF2 DAC5" }, + { "IF2 DAC7 Mux", "Slot6", "IF2 DAC6" }, + { "IF2 DAC7 Mux", "Slot7", "IF2 DAC7" }, + + { "IF2 DAC01", NULL, "IF2 DAC0 Mux" }, + { "IF2 DAC01", NULL, "IF2 DAC1 Mux" }, + { "IF2 DAC23", NULL, "IF2 DAC2 Mux" }, + { "IF2 DAC23", NULL, "IF2 DAC3 Mux" }, + { "IF2 DAC45", NULL, "IF2 DAC4 Mux" }, + { "IF2 DAC45", NULL, "IF2 DAC5 Mux" }, + { "IF2 DAC67", NULL, "IF2 DAC6 Mux" }, + { "IF2 DAC67", NULL, "IF2 DAC7 Mux" }, { "IF3 DAC", NULL, "AIF3RX" }, { "IF3 DAC", NULL, "I2S3" }, diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 9d473b2..2979d5a 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -799,7 +799,7 @@ #define RT5677_PDM2_I2C_EXE (0x1 << 1) #define RT5677_PDM2_I2C_BUSY (0x1 << 0) -/* MX3B TDM1 control 1 (0x3b) */ +/* TDM1 control 1 (0x3b) */ #define RT5677_IF1_ADC_MODE_MASK (0x1 << 12) #define RT5677_IF1_ADC_MODE_SFT 12 #define RT5677_IF1_ADC_MODE_I2S (0x0 << 12) @@ -813,7 +813,7 @@ #define RT5677_IF1_ADC4_SWAP_MASK (0x3 << 0) #define RT5677_IF1_ADC4_SWAP_SFT 0 -/* MX3C TDM1 control 2 (0x3c) */ +/* TDM1 control 2 (0x3c) */ #define RT5677_IF1_ADC4_MASK (0x3 << 10) #define RT5677_IF1_ADC4_SFT 10 #define RT5677_IF1_ADC3_MASK (0x3 << 8) @@ -825,7 +825,27 @@ #define RT5677_IF1_ADC_CTRL_MASK (0x7 << 0) #define RT5677_IF1_ADC_CTRL_SFT 0 -/* MX40 TDM2 control 1 (0x40) */ +/* TDM1 control 4 (0x3e) */ +#define RT5677_IF1_DAC0_MASK (0x7 << 12) +#define RT5677_IF1_DAC0_SFT 12 +#define RT5677_IF1_DAC1_MASK (0x7 << 8) +#define RT5677_IF1_DAC1_SFT 8 +#define RT5677_IF1_DAC2_MASK (0x7 << 4) +#define RT5677_IF1_DAC2_SFT 4 +#define RT5677_IF1_DAC3_MASK (0x7 << 0) +#define RT5677_IF1_DAC3_SFT 0 + +/* TDM1 control 5 (0x3f) */ +#define RT5677_IF1_DAC4_MASK (0x7 << 12) +#define RT5677_IF1_DAC4_SFT 12 +#define RT5677_IF1_DAC5_MASK (0x7 << 8) +#define RT5677_IF1_DAC5_SFT 8 +#define RT5677_IF1_DAC6_MASK (0x7 << 4) +#define RT5677_IF1_DAC6_SFT 4 +#define RT5677_IF1_DAC7_MASK (0x7 << 0) +#define RT5677_IF1_DAC7_SFT 0 + +/* TDM2 control 1 (0x40) */ #define RT5677_IF2_ADC_MODE_MASK (0x1 << 12) #define RT5677_IF2_ADC_MODE_SFT 12 #define RT5677_IF2_ADC_MODE_I2S (0x0 << 12) @@ -839,7 +859,7 @@ #define RT5677_IF2_ADC4_SWAP_MASK (0x3 << 0) #define RT5677_IF2_ADC4_SWAP_SFT 0 -/* MX41 TDM2 control 2 (0x41) */ +/* TDM2 control 2 (0x41) */ #define RT5677_IF2_ADC4_MASK (0x3 << 10) #define RT5677_IF2_ADC4_SFT 10 #define RT5677_IF2_ADC3_MASK (0x3 << 8) @@ -851,6 +871,26 @@ #define RT5677_IF2_ADC_CTRL_MASK (0x7 << 0) #define RT5677_IF2_ADC_CTRL_SFT 0 +/* TDM2 control 4 (0x43) */ +#define RT5677_IF2_DAC0_MASK (0x7 << 12) +#define RT5677_IF2_DAC0_SFT 12 +#define RT5677_IF2_DAC1_MASK (0x7 << 8) +#define RT5677_IF2_DAC1_SFT 8 +#define RT5677_IF2_DAC2_MASK (0x7 << 4) +#define RT5677_IF2_DAC2_SFT 4 +#define RT5677_IF2_DAC3_MASK (0x7 << 0) +#define RT5677_IF2_DAC3_SFT 0 + +/* TDM2 control 5 (0x44) */ +#define RT5677_IF2_DAC4_MASK (0x7 << 12) +#define RT5677_IF2_DAC4_SFT 12 +#define RT5677_IF2_DAC5_MASK (0x7 << 8) +#define RT5677_IF2_DAC5_SFT 8 +#define RT5677_IF2_DAC6_MASK (0x7 << 4) +#define RT5677_IF2_DAC6_SFT 4 +#define RT5677_IF2_DAC7_MASK (0x7 << 0) +#define RT5677_IF2_DAC7_SFT 0 + /* Digital Microphone Control 1 (0x50) */ #define RT5677_DMIC_1_EN_MASK (0x1 << 15) #define RT5677_DMIC_1_EN_SFT 15 -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] ALC1150 COEF
by Kailang 12 Nov '14

12 Nov '14

I merge all codec into alc_fill_eapd_coef(). Please see attach. Many Thanks. ^^ > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, November 12, 2014 4:39 PM > To: Kailang > Subject: Re: ALC1150 COEF > > At Wed, 12 Nov 2014 08:27:56 +0000, > Kailang wrote: > > > > Hi Takashi, > > > > I cooked a patch for this issue as attach. > > It is not include ALC26x ALC28x ALC29x. > > If you agree this patch, > > could it merge into one patch? > > Yes, it looks good. Could you submit it with proper > changelog etc as usual to alsa-devel ML? Then I'll queue it > for 3.18-rc5. > > > thanks, > > Takashi > > > > > BR, > > Kailang > > > > > -----Original Message----- > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > Sent: Wednesday, November 12, 2014 4:08 AM > > > To: Kailang > > > Subject: ALC1150 COEF > > > > > > Hi Kailang, > > > > > > we've got bug reports about the silent output on mobos > with ALC1150 > > > after Windows reboot. It seems that COEF 0x07 bit > > > 0x02 plays some role. Could you check it? > > > https://bugzilla.kernel.org/show_bug.cgi?id=87771 > > > > > > thanks, > > > > > > Takashi > > > > > > ------Please consider the environment before printing this e-mail. > > > > > [2 0000-reset-eapd-control.patch <application/octet-stream > (base64)>] > > >

2 1

[alsa-devel] [PATCH 0/4] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 12 Nov '14

12 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Krishna Mohan Dani (4): ASoC: ALC5631/RT5631: Add device tree binding documentation ASoC: Samsung: Add arndale_rt5631 machine driver Sound: Kconfig: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 48 ++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 11 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 166 ++++++++++++++++++++ 6 files changed, 235 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

2 5

[alsa-devel] [PATCH 1/4] ASoC: rt5670: make bias level more reasonable
by bardliao＠realtek.com 12 Nov '14

12 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c4c42d5..85dc210 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2309,6 +2309,8 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2330,16 +2332,27 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); - snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); - snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); - snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); - snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); snd_soc_update_bits(codec, RT5670_PWR_ANLG1, RT5670_LDO_SEL_MASK, 0x1); break; + case SND_SOC_BIAS_OFF: + if (rt5670->pdata.jd_mode) + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_MB | RT5670_PWR_BG); + else + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + break; default: break; -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix memory leak in FTU quirk
by Takashi Iwai 11 Nov '14

11 Nov '14

M-audio FastTrack Ultra quirk doesn't release the kzalloc'ed memory. This patch adds the private_free callback to release it properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: free the correct field sound/usb/mixer_quirks.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f119a41ed9a9..7c83bab69dee 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -885,6 +885,11 @@ static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, return changed; } +static void kctl_private_value_free(struct snd_kcontrol *kctl) +{ + kfree((void *)kctl->private_value); +} + static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, int validx, int bUnitID) { @@ -919,6 +924,7 @@ static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, return -ENOMEM; } + kctl->private_free = kctl_private_value_free; err = snd_ctl_add(mixer->chip->card, kctl); if (err < 0) return err; -- 2.1.3

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Fix memory leak in FTU quirk
by Takashi Iwai 11 Nov '14

11 Nov '14

M-audio FastTrack Ultra quirk doesn't release the kzalloc'ed memory. This patch adds the private_free callback to release it properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 45203168ac6d..f6f8acf0446b 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1021,6 +1021,7 @@ static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, return -ENOMEM; } + kctl->private_free = usb_mixer_elem_free; err = snd_ctl_add(mixer->chip->card, kctl); if (err < 0) return err; -- 2.1.3

1 1

[alsa-devel] [PATCH v6 0/3] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 11 Nov '14

11 Nov '14

This version of patchset contains following style changes since v5: - nodes sorted in alphabetical order; - pinctrl entries sorted alphabetically; - new line inserted between nodes.

2 4

[alsa-devel] [PATCH 0/2] Subject: [PATCH 0/2] ASoC: tlv320aic3x: soc_enum array conversion and pop reduction
by Peter Ujfalusi 11 Nov '14

11 Nov '14

Hi, Convert the soc_enum arrays to individual items and add the pop reduction patch on top of this change. Regards, Peter --- Misael Lopez Cruz (1): ASoC: tlv320aic3x: Add output driver pop reduction controls Peter Ujfalusi (1): ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual sound/soc/codecs/tlv320aic3x.c | 165 +++++++++++++++++++++++------------------ 1 file changed, 93 insertions(+), 72 deletions(-) -- 2.1.3

2 3

[alsa-devel] Realtek HDA SKU/assembly parsing
by Daniel Drake 11 Nov '14

11 Nov '14

Hi, I see that there is a function that parses the subsystem ID for various realtek codecs, alc_auto_parse_customize_define Is there a public specification that documents the specific meaning of each bit here? Thanks Daniel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH] ASoC: rt5645: remove unused rt5645_clk_sel_put
by Bard Liao 11 Nov '14

11 Nov '14

Remove rt5645_clk_sel_put function since it is never used. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 59 ----------------------------------------------- 1 file changed, 59 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1adab38..d6298b3 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -443,65 +443,6 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); -static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - unsigned int u_bit = 0, p_bit = 0; - struct soc_enum *em = - (struct soc_enum *)kcontrol->private_value; - - switch (em->reg) { - case RT5645_ASRC_2: - switch (em->shift_l) { - case 0: - u_bit = 0x8; - p_bit = RT5645_PWR_ADC_S1F; - break; - case 4: - u_bit = 0x100; - p_bit = RT5645_PWR_DAC_MF_R; - break; - case 8: - u_bit = 0x200; - p_bit = RT5645_PWR_DAC_MF_L; - break; - case 12: - u_bit = 0x400; - p_bit = RT5645_PWR_DAC_S1F; - break; - } - break; - case RT5645_ASRC_3: - switch (em->shift_l) { - case 0: - u_bit = 0x1; - p_bit = RT5645_PWR_ADC_MF_R; - break; - case 4: - u_bit = 0x2; - p_bit = RT5645_PWR_ADC_MF_L; - break; - } - break; - } - - if (u_bit || p_bit) { - switch (ucontrol->value.integer.value[0]) { - case 1 ... 4: /*enable*/ - if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) - snd_soc_update_bits(codec, - RT5645_ASRC_1, u_bit, u_bit); - break; - default: /*disable*/ - snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); - break; - } - } - - return snd_soc_put_enum_double(kcontrol, ucontrol); -} - static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: rt5670: add rt5672 codec support
by bardliao＠realtek.com 11 Nov '14

11 Nov '14

From: Bard Liao <bardliao(a)realtek.com> rt5672 is very similar to rt5670. Therefore we use one codec driver to support both codecs. The difference between rt5670 and rt5672 is there is some difference in their dapm routing table. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Add { "rt5671", 0 }, in rt5670_i2c_id[] --- sound/soc/codecs/rt5670.c | 71 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/rt5670.h | 6 ++++ 2 files changed, 65 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9bd8b4f..c4c42d5 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1595,29 +1595,40 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { /* PDM */ SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, RT5670_PWR_PDM1_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, - RT5670_PWR_PDM2_BIT, 0, NULL, 0), SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), - SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), - SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), /* Output Lines */ SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), SND_SOC_DAPM_OUTPUT("PDM1L"), SND_SOC_DAPM_OUTPUT("PDM1R"), SND_SOC_DAPM_OUTPUT("PDM2L"), SND_SOC_DAPM_OUTPUT("PDM2R"), }; +static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = { + SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), +}; + static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1970,12 +1981,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, { "PDM1 R Mux", NULL, "PDM1 Power" }, - { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, - { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, - { "PDM2 L Mux", NULL, "PDM2 Power" }, - { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, - { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, - { "PDM2 R Mux", NULL, "PDM2 Power" }, { "HP Amp", NULL, "HPO MIX" }, { "HP Amp", NULL, "Mic Det Power" }, @@ -1993,13 +1998,30 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "LOUTR", NULL, "LOUT R Playback" }, { "LOUTL", NULL, "Improve HP Amp Drv" }, { "LOUTR", NULL, "Improve HP Amp Drv" }, +}; +static const struct snd_soc_dapm_route rt5670_specific_dapm_routes[] = { + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, { "PDM1L", NULL, "PDM1 L Mux" }, { "PDM1R", NULL, "PDM1 R Mux" }, { "PDM2L", NULL, "PDM2 L Mux" }, { "PDM2R", NULL, "PDM2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5672_specific_dapm_routes[] = { + { "SPO Amp", NULL, "PDM1 L Mux" }, + { "SPO Amp", NULL, "PDM1 R Mux" }, + { "SPOLP", NULL, "SPO Amp" }, + { "SPOLN", NULL, "SPO Amp" }, + { "SPORP", NULL, "SPO Amp" }, + { "SPORN", NULL, "SPO Amp" }, +}; + static int rt5670_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2331,6 +2353,29 @@ static int rt5670_probe(struct snd_soc_codec *codec) { struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (snd_soc_read(codec, RT5670_RESET) & RT5670_ID_MASK) { + case RT5670_ID_5670: + case RT5670_ID_5671: + snd_soc_dapm_new_controls(&codec->dapm, + rt5670_specific_dapm_widgets, + ARRAY_SIZE(rt5670_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5670_specific_dapm_routes, + ARRAY_SIZE(rt5670_specific_dapm_routes)); + break; + case RT5670_ID_5672: + snd_soc_dapm_new_controls(&codec->dapm, + rt5672_specific_dapm_widgets, + ARRAY_SIZE(rt5672_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5672_specific_dapm_routes, + ARRAY_SIZE(rt5672_specific_dapm_routes)); + break; + default: + dev_err(codec->dev, + "The driver is for RT5670 RT5671 or RT5672 only\n"); + return -ENODEV; + } rt5670->codec = codec; return 0; @@ -2452,6 +2497,8 @@ static const struct regmap_config rt5670_regmap = { static const struct i2c_device_id rt5670_i2c_id[] = { { "rt5670", 0 }, + { "rt5671", 0 }, + { "rt5672", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index a0b5c85..d11b9c2 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -228,6 +228,12 @@ #define RT5670_R_VOL_MASK (0x3f) #define RT5670_R_VOL_SFT 0 +/* SW Reset & Device ID (0x00) */ +#define RT5670_ID_MASK (0x3 << 1) +#define RT5670_ID_5670 (0x0 << 1) +#define RT5670_ID_5672 (0x1 << 1) +#define RT5670_ID_5671 (0x2 << 1) + /* Combo Jack Control 1 (0x0a) */ #define RT5670_CBJ_BST1_MASK (0xf << 12) #define RT5670_CBJ_BST1_SFT (12) -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/6] ASoC: davinci-mcasp: fixes and over/underrun handling
by Peter Ujfalusi 11 Nov '14

11 Nov '14

Hi, Some fixes for the mcasp code: - rx formating fix - constraint for number of channels and sample bits handling of over/underrun: When such event happens (in McASP vs DMA level) we force restart the stream in order to get McASP and the DMA in sync again. Regards, Peter --- Misael Lopez Cruz (2): ASoC: davinci-mcasp: Active slots depend on active serializers ASoC: davinci-mcasp: Add overrun/underrun event handling Peter Ujfalusi (4): ASoC: davinci-mcasp: Validate tdm_slots parameter at probe time ASoC: davinci-mcasp: Place constraint on number of channels ASoC: davinci-mcasp: Symmetric sample bits for IIS mode ASoC: davinci-mcasp: Fix rx format when more bclk is used on the bus .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 212 +++++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 10 + 3 files changed, 210 insertions(+), 14 deletions(-) -- 2.1.3

2 14

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Add output driver pop reduction controls
by Peter Ujfalusi 11 Nov '14

11 Nov '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Output driver has two parameters that can be configured to reduce pop noise: power-on delay and ramp-up step time. Two new kcontrols have been added to set these parameters. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f7c2a575a892..70f8b8be9173 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -270,6 +270,15 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), }; +static const char * const aic3x_poweron_time[] = { + "0us", "10us", "100us", "1ms", "10ms", "50ms", + "100ms", "200ms", "400ms", "800ms", "2s", "4s" }; +static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" }; +static const struct soc_enum aic3x_pop_reduction_enum[] = { + SOC_ENUM_SINGLE(HPOUT_POP_REDUCTION, 4, 12, aic3x_poweron_time), + SOC_ENUM_SINGLE(HPOUT_POP_REDUCTION, 2, 4, aic3x_rampup_step), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -399,6 +408,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + /* Pop reduction */ + SOC_ENUM("Output Driver Power-On time", aic3x_pop_reduction_enum[0]), + SOC_ENUM("Output Driver Ramp-up step", aic3x_pop_reduction_enum[1]), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { -- 2.1.3

2 6

[alsa-devel] [PATCH 0/2 v6] Digidesign Mbox 1 enhancements
by Damien Zammit 11 Nov '14

11 Nov '14

Revision 6. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 ++++++++++++---- 3 files changed, 171 insertions(+), 10 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: Fix kerneldoc errors
by Takashi Iwai 11 Nov '14

11 Nov '14

Lots of typos and missing parameter descriptions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 11 ++++++++--- sound/soc/soc-core.c | 21 ++++++++++++++------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-devres.c | 2 +- sound/soc/soc-jack.c | 2 +- 5 files changed, 25 insertions(+), 13 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130037a0..25d0bdbbeb11 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -583,6 +583,7 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, /** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * + * @list: linked list of assigned pins to the same jack * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported @@ -602,7 +603,7 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container + * @list: list container */ struct snd_soc_jack_zone { unsigned int min_mv; @@ -618,16 +619,20 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debouce time in ms * @wake: enable as wake source + * @jack: assigned jack + * @work: delayed_work for GPIO detection + * @desc: GPIO descriptor * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an * ADC). + * @data: arbitrary data passed to jack_status_check callback */ struct snd_soc_jack_gpio { unsigned int gpio; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a23a0e9..74a8a80e2d9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2113,6 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: AC97 opts to set + * @pdev: platform to deal with * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. @@ -2685,7 +2687,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -3458,7 +3460,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -4082,6 +4084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core + * @dev: device to unregister * */ void snd_soc_unregister_component(struct device *dev) @@ -4119,7 +4122,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -4158,7 +4161,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: platform to register + * @platform_drv: the driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -4215,7 +4219,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -4306,7 +4310,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: codec to register + * @codec_drv: the codec driver to register + * @dai_drv: the array of DAI drivers to register + * @num_dai: the number of DAIs to register */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -4406,7 +4413,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..7050ec1d5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2680,7 +2680,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card object containing widgets * * Checks the codec for any new dapm widgets and creates them if found. * diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..ef1d42d7c6f6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- 2.1.3

2 8

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add ASRC support
by bardliao＠realtek.com 11 Nov '14

11 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch add ASRC support for rt5645 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 144 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 144 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc442f4..3131e91 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -443,6 +443,65 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); +static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5645_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5645_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5645_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5645_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5645_PWR_DAC_S1F; + break; + } + break; + case RT5645_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5645_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5645_PWR_ADC_MF_L; + break; + } + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5645_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -554,6 +613,53 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5645_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5645_ASRC_3; + shift = 4; + break; + case 3: + reg = RT5645_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5645_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5645_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5645_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, @@ -1246,6 +1352,30 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL, RT5645_PWR_MIC_DET_BIT, 0, NULL, 0), + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5645_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5645_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5645_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5645_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5645_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5645_ASRC_1, + 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5645_ASRC_1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5645_ASRC_1, + 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5645_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5645_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5645_ASRC_1, + 0, 0, NULL, 0), + /* Input Side */ /* micbias */ SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2, @@ -1504,6 +1634,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { + { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "dac mono right filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + { "IN1P", NULL, "LDO2" }, { "IN2P", NULL, "LDO2" }, @@ -1550,12 +1691,15 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC" }, { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC" }, { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC" }, { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, -- 1.8.1.1.439.g50a6b54

4 5

[alsa-devel] Issue with hda_intel on i7+Z97 chipset
by Daniel Silverstone 10 Nov '14

10 Nov '14

Hi, Firstly, I hope this is a suitable way to report this issue. I am using Debian/Wheezy on a reasonably recent i7 based system which has a Z97 motherboard chipset. In Windows, sound is perfect, but in Linux it's very crackly (direct through alsa) and if I bring pulseaudio into the mix then it reports buffer/interrupt issues on the microphone along the lines of: Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Oct 19 16:03:33 listless pulseaudio[3780]: [alsa-source] alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Similar messages are shown for issues with playback: user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. user.log.2.gz:Oct 18 21:50:25 listless pulseaudio[3780]: [alsa-sink] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. In both cases, sound exhibts crackles, pops, crunches, stalls, repeats etc. I can mitigate the issue by working one or more of the CPU cores hard with bzip2 -c </dev/urandom > /dev/null Of course, this makes my system fans run harder etc, which is a bit of a pain for me, and it's only masking the issue, not solving it. For reference, my motherboard's dmi shows: Manufacturer: Gigabyte Technology Co., Ltd. Product Name: Z97-D3H-CF And the CPU reports as: Family: Core i7 Manufacturer: Intel ID: C3 06 03 00 FF FB EB BF Signature: Type 0, Family 6, Model 60, Stepping 3 Version: Intel(R) Core(TM) i7-4790K CPU @ 4.00GHz The relevant PCI devices are: 00:03.0 0403: 8086:0c0c (rev 06) Subsystem: 8086:2010 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 51 Region 0: Memory at f7834000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel and 00:1b.0 0403: 8086:8ca0 Subsystem: 1458:a182 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at f7830000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel and it's possible that this kernel message is relevant: [ 17.248493] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [ 17.251735] hda-intel: Invalid position buffer, using LPIB read method instead. I have tried a number of bdl_pos_adj between 1 and 32 and I have tried enable_msi=1 Neither seem to have helped significantly. I am currently running 3.12-0.bpo.1-amd64 from Debian's wheezy-backports repository. Unfortunately anything newer seems to lock up during boot and I've not managed to track down what or why as yet. I was hoping that you may have some idea of what is going on and either some suggestions for improving my situation on 3.12 or else some desire for further debug data such that a newer kernel (once I can persuade one to boot) may perform better for me. Please let me know if you require further details from me in order to address this. Regards, Daniel. -- Daniel Silverstone http://www.digital-scurf.org/ PGP mail accepted and encouraged. Key Id: 3CCE BABE 206C 3B69

1 1

[alsa-devel] [PATCH 0/3] ASoC: twl4030: support routine to external VOICE source.
by NeilBrown 10 Nov '14

10 Nov '14

The 'voice' port of the twl4030 on my board is connected to a GSM modem, not to the CPU. As such it is not visible to ASLA and normal approaches to configuring the interface (as one end of a DAI) don't apply. I need a way to tell the twl4030 that the connected device will be master of 'clk' and 'FRM', and whether they are inverted. Using device tree seems the correct approach - I am describing the properties of the hardware attached the 'voice' port. Using magic numbers from include/sound/soc-dai.h almost certainly is not the right way to represent this information, but it isn't clear to me what a good way would be. So the current patch uses magic numbers as a proof of concept, and I'm asking if there is any establish precedent, or any suggestions on how to record in device tree the 'master' and 'polarity' of CLK and FRM signals for an external device. [I currently use (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS)] This is in the second patch. The first silences a pointless warning (when it is pointless) and the last allowed the voice linked to be turned on or off via ALSA controls. If the first and last can be applied without finalizing the device-tree part yet, that would be great. Thanks, NeilBrown --- NeilBrown (3): ASoC: twl4030: don't report EBUSY if no change requested. ASoC: twl4030: allow voice port to be connected externally. ASoC: twl4030: enable routing audio to 'voice' interface. .../devicetree/bindings/mfd/twl4030-audio.txt | 7 ++ include/linux/i2c/twl.h | 3 + sound/soc/codecs/twl4030.c | 78 ++++++++++++++++++-- 3 files changed, 82 insertions(+), 6 deletions(-) -- Signature

4 15

[alsa-devel] [PATCH] ASoC: mxs: mxs-saif: Register the irq with the device name
by Fabio Estevam 10 Nov '14

10 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Instead of registering the irq name with the driver name, it's better to pass the device name so that we have a more explicit indication as to what saif instance the irq is related: $ cat /proc/interrupts CPU0 ... 214: 4 - 59 80042000.saif 215: 0 - 58 80046000.saif Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index eade2742..d986508 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -773,7 +773,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->dev = &pdev->dev; ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, - "mxs-saif", saif); + dev_name(&pdev->dev), saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); return ret; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/2] Fix some 'const' problems
by Jean-Francois Moine 10 Nov '14

10 Nov '14

A cast was often needed to call some OF functions, removing a 'const' attribute. This patchset removes 'const' from the device_node pointers of the sound/soc structures and also removes some useless casts in the sound/soc tree. v2: remove const from the pointers instead of modifying the OF functions (Russell King) Jean-Francois Moine (2): ASoC: Remove 'const' from the device_node pointers ASoC: simple-card: Remove useless casts include/sound/soc.h | 6 +++--- sound/soc/generic/simple-card.c | 7 ++----- sound/soc/samsung/odroidx2_max98090.c | 4 ++-- sound/soc/ux500/mop500.c | 6 ++---- 4 files changed, 9 insertions(+), 14 deletions(-) -- 2.1.3

3 8

[alsa-devel] [PATCH 0/8] ASoC: Replace remaining users of the CODEC mutex
by Lars-Peter Clausen 10 Nov '14

10 Nov '14

We've stopped using the snd_soc_codec mutex in the ASoC core itself a while ago. There is still a handful of drivers which use the mutex to protect some of their data structures. This patch moves the lock for the remaining users to the drivers private data struct and then removes it from the snd_soc_codec struct. - Lars Lars-Peter Clausen (8): ASoC: ab8500-codec: Move control lock to the driver level ASoC: max98095: Move mutex to the driver level ASoC: wm5102: Move ultrasonic response settings lock to the driver level ASoC: wm8731: Move the deemph lock to the driver level ASoC: wm8903: Move the deemph lock to the driver level ASoC: wm8958: Move DSP firmware lock to driver level ASoC: wm8962: Move DSP enable lock to the driver level ASoC: Remove CODEC mutex include/linux/mfd/arizona/core.h | 1 + include/sound/soc.h | 1 - sound/soc/codecs/ab8500-codec.c | 32 ++++++++++++++++---------------- sound/soc/codecs/arizona.c | 4 ++-- sound/soc/codecs/max98095.c | 12 ++++++++---- sound/soc/codecs/wm5102.c | 16 ++++++++-------- sound/soc/codecs/wm8731.c | 9 +++++++-- sound/soc/codecs/wm8903.c | 8 ++++++-- sound/soc/codecs/wm8958-dsp2.c | 12 ++++++------ sound/soc/codecs/wm8962.c | 8 ++++++-- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm8994.h | 2 ++ sound/soc/soc-core.c | 1 - 13 files changed, 64 insertions(+), 44 deletions(-) -- 1.8.0

3 10

[alsa-devel] [PATCH 0/4] Yet more kenreldoc fixes
by Takashi Iwai 10 Nov '14

10 Nov '14

Hi, these are a series of patches to fix kerneldoc errors / warnings. These aren't actaully compiled from DocBook directory, so they are harmless, but better to be fixed, of course. [PATCH 1/4] ALSA: vx: Fix missing kenreldoc parameter descriptions [PATCH 2/4] ALSA: hda - Fix kerneldoc errors in patch_ca0132.c [PATCH 3/4] ALSA: mixart: Fix kerneldoc comments [PATCH 4/4] ALSA: Fix invalid kerneldoc markers Takashi

2 8

[alsa-devel] [PATCH 0/5 v2] ASoC: rsnd: Add Volume Ramp support
by Kuninori Morimoto 10 Nov '14

10 Nov '14

Hi Mark These are v2 of Renesas R-Car sound Volume Ramp support. It uses "Volume Ramp Up/Down Rate" on last patch instead of "Period". Because it is good match naming, and it was difficult to calculate real period time on this HW. Kuninori Morimoto (5): ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: Add Volume Ramp support sound/soc/sh/rcar/dvc.c | 126 +++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/gen.c | 3 ++ sound/soc/sh/rcar/rsnd.h | 6 +++ 3 files changed, 116 insertions(+), 19 deletions(-) Best regards --- Kuninori Morimoto

2 21

[alsa-devel] [PATCH] ASoC: rsnd: fallback to PIO mode if DMA mode was failed
by Kuninori Morimoto 10 Nov '14

10 Nov '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current Renesas R-Car sound driver probe will be failed if it try to use DMA mode and it couldn't use for some reasons. But PIO mode works even though in such case. This patch try to fallback to PIO mode if DMA mode probing was failed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 74 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/sh/rcar/ssi.c | 15 ++++++++++ 2 files changed, 85 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5205618..110b99d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -349,7 +349,7 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_name); if (!dma->chan) { dev_err(dev, "can't get dma channel\n"); - return -EIO; + goto rsnd_dma_channel_err; } ret = dmaengine_slave_config(dma->chan, &cfg); @@ -363,8 +363,15 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, rsnd_dma_init_err: rsnd_dma_quit(priv, dma); +rsnd_dma_channel_err: - return ret; + /* + * DMA failed. try to PIO mode + * see + * rsnd_ssi_dma_remove() + * rsnd_rdai_continuance_probe() + */ + return -EAGAIN; } void rsnd_dma_quit(struct rsnd_priv *priv, @@ -456,6 +463,13 @@ static int rsnd_dai_connect(struct rsnd_mod *mod, return 0; } +static void rsnd_dai_disconnect(struct rsnd_mod *mod, + struct rsnd_dai_stream *io) +{ + mod->io = NULL; + io->mod[mod->type] = NULL; +} + int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) { int id = rdai - priv->rdai; @@ -686,6 +700,20 @@ static const struct snd_soc_dai_ops rsnd_soc_dai_ops = { ret; \ }) +#define rsnd_path_break(priv, io, type) \ +{ \ + struct rsnd_mod *mod; \ + int id = -1; \ + \ + if (rsnd_is_enable_path(io, type)) { \ + id = rsnd_info_id(priv, io, type); \ + if (id >= 0) { \ + mod = rsnd_##type##_mod_get(priv, id); \ + rsnd_dai_disconnect(mod, io); \ + } \ + } \ +} + static int rsnd_path_init(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) @@ -977,6 +1005,44 @@ static const struct snd_soc_component_driver rsnd_soc_component = { .name = "rsnd", }; +static int rsnd_rdai_continuance_probe(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + int is_play) +{ + struct rsnd_dai_stream *io = is_play ? &rdai->playback : &rdai->capture; + int ret; + + ret = rsnd_dai_call(probe, io, rdai); + if (ret == -EAGAIN) { + /* + * Fallback to PIO mode + */ + + /* + * call "remove" for SSI/SRC/DVC + * SSI will be switch to PIO mode if it was DMA mode + * see + * rsnd_dma_init() + * rsnd_ssi_dma_remove() + */ + rsnd_dai_call(remove, io, rdai); + + /* + * remove SRC/DVC from DAI, + */ + rsnd_path_break(priv, io, src); + rsnd_path_break(priv, io, dvc); + + /* + * retry to "probe". + * DAI has SSI which is PIO mode only now. + */ + ret = rsnd_dai_call(probe, io, rdai); + } + + return ret; +} + /* * rsnd probe */ @@ -1038,11 +1104,11 @@ static int rsnd_probe(struct platform_device *pdev) } for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(probe, &rdai->playback, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 1); if (ret) goto exit_snd_probe; - ret = rsnd_dai_call(probe, &rdai->capture, rdai); + ret = rsnd_rdai_continuance_probe(priv, rdai, 0); if (ret) goto exit_snd_probe; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index cae08b7..346d3dc 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -465,8 +465,23 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, static int rsnd_ssi_dma_remove(struct rsnd_mod *mod, struct rsnd_dai *rdai) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); + rsnd_dma_quit(rsnd_mod_to_priv(mod), rsnd_mod_to_dma(mod)); + /* + * fallback to PIO + * + * SSI .probe might be called again. + * see + * rsnd_rdai_continuance_probe() + */ + mod->ops = &rsnd_ssi_pio_ops; + + dev_info(dev, "%s[%d] fallback to PIO mode\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: rsnd: tidyup debug message format and timing
by Kuninori Morimoto 10 Nov '14

10 Nov '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current Renesas R-Car sound driver debug message is using random format (ex "ssi0: xxx" / "SSI0 xxx" / "ssi[0]: xxx") and confusable timing ("xxx probe failed" and "xxx probed" are shown in same time) This patch fixes these Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rcar/dvc.c | 3 ++- sound/soc/sh/rcar/gen.c | 6 +++--- sound/soc/sh/rcar/src.c | 11 +++++++---- sound/soc/sh/rcar/ssi.c | 25 ++++++++++++++++--------- 5 files changed, 29 insertions(+), 18 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1922ec5..5205618 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -409,7 +409,7 @@ u32 rsnd_get_adinr(struct rsnd_mod *mod) ({ \ struct rsnd_priv *priv = rsnd_mod_to_priv(mod); \ struct device *dev = rsnd_priv_to_dev(priv); \ - dev_dbg(dev, "%s [%d] %s\n", \ + dev_dbg(dev, "%s[%d] %s\n", \ rsnd_mod_name(mod), rsnd_mod_id(mod), #func); \ (mod)->ops->func(mod, rdai); \ }) diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index d6d8e64..e2c8473 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -141,7 +141,8 @@ static int rsnd_dvc_probe_gen2(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "%s (Gen2) is probed\n", rsnd_mod_name(mod)); + dev_dbg(dev, "%s[%d] (Gen2) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return 0; } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 4cb3202..87a6f2d 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -78,7 +78,7 @@ u32 rsnd_read(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return 0; - dev_dbg(dev, "r %s(%d) - %4d : %08x\n", + dev_dbg(dev, "r %s[%d] - %4d : %08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, val); regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); @@ -96,7 +96,7 @@ void rsnd_write(struct rsnd_priv *priv, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "w %s(%d) - %4d : %08x\n", + dev_dbg(dev, "w %s[%d] - %4d : %08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data); regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); @@ -111,7 +111,7 @@ void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, if (!rsnd_is_accessible_reg(priv, gen, reg)) return; - dev_dbg(dev, "b %s(%d) - %4d : %08x/%08x\n", + dev_dbg(dev, "b %s[%d] - %4d : %08x/%08x\n", rsnd_mod_name(mod), rsnd_mod_id(mod), reg, data, mask); regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 9183e01..4679501 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -438,7 +438,8 @@ static int rsnd_src_probe_gen1(struct rsnd_mod *mod, struct rsnd_priv *priv = rsnd_mod_to_priv(mod); struct device *dev = rsnd_priv_to_dev(priv); - dev_dbg(dev, "%s (Gen1) is probed\n", rsnd_mod_name(mod)); + dev_dbg(dev, "%s[%d] (Gen1) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return 0; } @@ -578,9 +579,11 @@ static int rsnd_src_probe_gen2(struct rsnd_mod *mod, rsnd_info_is_playback(priv, src), src->info->dma_id); if (ret < 0) - dev_err(dev, "SRC DMA failed\n"); - - dev_dbg(dev, "%s (Gen2) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (Gen2) failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (Gen2) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 34e8400..cae08b7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -159,7 +159,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_ssi *ssi, ssi->cr_clk = FORCE | SWL_32 | SCKD | SWSD | CKDV(j); - dev_dbg(dev, "ssi%d outputs %u Hz\n", + dev_dbg(dev, "%s[%d] outputs %u Hz\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod), rate); return 0; @@ -206,7 +207,8 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, ssi->usrcnt++; - dev_dbg(dev, "ssi%d hw started\n", rsnd_mod_id(&ssi->mod)); + dev_dbg(dev, "%s[%d] hw started\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); } static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, @@ -249,7 +251,8 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, clk_disable_unprepare(ssi->clk); } - dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); + dev_dbg(dev, "%s[%d] hw stopped\n", + rsnd_mod_name(&ssi->mod), rsnd_mod_id(&ssi->mod)); } /* @@ -385,9 +388,11 @@ static int rsnd_ssi_pio_probe(struct rsnd_mod *mod, IRQF_SHARED, dev_name(dev), ssi); if (ret) - dev_err(dev, "SSI request interrupt failed\n"); - - dev_dbg(dev, "%s (PIO) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (PIO) request interrupt failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (PIO) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } @@ -448,9 +453,11 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, dma_id); if (ret < 0) - dev_err(dev, "SSI DMA failed\n"); - - dev_dbg(dev, "%s (DMA) is probed\n", rsnd_mod_name(mod)); + dev_err(dev, "%s[%d] (DMA) is failed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + else + dev_dbg(dev, "%s[%d] (DMA) is probed\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); return ret; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/4] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 10 Nov '14

10 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Krishna Mohan Dani (4): ASoC: ALC5631: Add device tree binding documentation Sound: Samsung: Add sound card machine driver for Arndale Board Sound: Kconfig: Adding the description of the codec Sound: DT: Adding Device Tree compatibility to Realtek's ALC5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 41 +++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 11 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 167 ++++++++++++++++++++ 6 files changed, 229 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

3 7

[alsa-devel] [PATCH] ASoC: davinvi-mcasp: Balance pm_runtime_enable() on probe failure
by Anil Kumar 10 Nov '14

10 Nov '14

If probe fails then we need to call pm_runtime_disable() to balance out the previous pm_runtime_enable() call. Signed-off-by: Anil Kumar <anilk4.v(a)gmail.com> --- :100644 100644 0eed9b1... 99c747d... M sound/soc/davinci/davinci-mcasp.c sound/soc/davinci/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0eed9b1..99c747d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1235,6 +1235,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = pm_runtime_get_sync(&pdev->dev); if (IS_ERR_VALUE(ret)) { dev_err(&pdev->dev, "pm_runtime_get_sync() failed\n"); + pm_runtime_disable(&pdev->dev); return ret; } -- 1.9.2

3 2

[alsa-devel] sound: enable sound support on n900 on devicetree-based boot
by Pavel Machek 10 Nov '14

10 Nov '14

With device tree, it is possible (and encouraged) to build N900 kernels without CONFIG_MACH_NOKIA_RX51. Update config file to enable the driver build in this case. This makes sound work on my n900 under 3.18-rc1. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index d44463a..0d7b707 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) && I2C + depends on SND_OMAP_SOC && ARM && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

2 4

[alsa-devel] [PATCH 0/2 v5] Digidesign Mbox 1 enhancements
by Damien Zammit 10 Nov '14

10 Nov '14

This is revision 5. I made a mistake interpreting your reviews, sorry guys. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 140 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 +++++++++++---- 3 files changed, 182 insertions(+), 10 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: korg1212: cleanup of printk
by Sudip Mukherjee 10 Nov '14

10 Nov '14

replaced all references of the debug messages via printk with dev_* macro (mostly dev_dbg). one reference was changed to pr_err as there the card might have been uninitialized. while compiling after these modifications, we are getting three warnings in function snd_korg1212_create: 2301: warning: format "%d" expects argument of type "int", but argument 6 has type "long unsigned int" [-Wformat] 2376: warning: format "%x" expects argument of type "unsigned int", but argument 5 has type "dma_addr_t" [-Wformat] 2376: warning: format "%d" expects argument of type "int", but argument 6 has type "size_t" [-Wformat] these warnings were prviously not visible as K1212_DEBUG_LEVEL was defined as 0. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/korg1212/korg1212.c | 406 +++++++++++++++++++++++------------------- 1 file changed, 219 insertions(+), 187 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 59d21c9..ab8e805 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -39,21 +39,6 @@ #include <asm/io.h> // ---------------------------------------------------------------------------- -// Debug Stuff -// ---------------------------------------------------------------------------- -#define K1212_DEBUG_LEVEL 0 -#if K1212_DEBUG_LEVEL > 0 -#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK(fmt,...) -#endif -#if K1212_DEBUG_LEVEL > 1 -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,...) -#endif - -// ---------------------------------------------------------------------------- // Record/Play Buffer Allocation Method. If K1212_LARGEALLOC is defined all // buffers are alocated as a large piece inside KorgSharedBuffer. // ---------------------------------------------------------------------------- @@ -530,12 +515,12 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, int rc = K1212_CMDRET_Success; if (!korg1212->outDoorbellPtr) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: CardUninitialized\n"); + pr_err("snd_korg1212_Send1212Command: Card Uninitialized\n"); return K1212_CMDRET_CardUninitialized; } - K1212_DEBUG_PRINTK("K1212_DEBUG: Card <- 0x%08x 0x%08x [%s]\n", - doorbellVal, mailBox0Val, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Card <- 0x%08x 0x%08x [%s]\n", + doorbellVal, mailBox0Val, stateName[korg1212->cardState]); for (retryCount = 0; retryCount < MAX_COMMAND_RETRIES; retryCount++) { writel(mailBox3Val, korg1212->mailbox3Ptr); writel(mailBox2Val, korg1212->mailbox2Ptr); @@ -562,7 +547,7 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, mailBox3Lo = readl(korg1212->mailbox3Ptr); if (mailBox3Lo & COMMAND_ACK_MASK) { if ((mailBox3Lo & DOORBELL_VAL_MASK) == (doorbellVal & DOORBELL_VAL_MASK)) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- Success\n"); + dev_dbg(korg1212->card->dev, "Card <- Success\n"); rc = K1212_CMDRET_Success; break; } @@ -571,7 +556,7 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, korg1212->cmdRetryCount += retryCount; if (retryCount >= MAX_COMMAND_RETRIES) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- NoAckFromCard\n"); + dev_err(korg1212->card->dev, "Card <- NoAckFromCard\n"); rc = K1212_CMDRET_NoAckFromCard; } @@ -612,20 +597,21 @@ static void snd_korg1212_timer_func(unsigned long data) korg1212->stop_pending_cnt = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Stop ack'ed [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop ack'ed [%s]\n", + stateName[korg1212->cardState]); } else { if (--korg1212->stop_pending_cnt > 0) { /* reprogram timer */ korg1212->timer.expires = jiffies + 1; add_timer(&korg1212->timer); } else { - snd_printd("korg1212_timer_func timeout\n"); + dev_err(korg1212->card->dev, + "korg1212_timer_func timeout\n"); korg1212->sharedBufferPtr->cardCommand = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK("K1212_DEBUG: Stop timeout [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop timeout [%s]\n", + stateName[korg1212->cardState]); } } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -660,8 +646,8 @@ static inline void snd_korg1212_setCardState(struct snd_korg1212 * korg1212, enu static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: OpenCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "K1212_DEBUG: OpenCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (korg1212->opencnt++ == 0) { snd_korg1212_TurnOffIdleMonitor(korg1212); @@ -674,8 +660,8 @@ static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "CloseCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (--(korg1212->opencnt)) { @@ -687,8 +673,9 @@ static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_StopPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "CloseCard - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { mutex_unlock(&korg1212->open_mutex); return 0; @@ -711,8 +698,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->setcnt); + dev_dbg(korg1212->card->dev, "SetupForPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->setcnt); if (korg1212->setcnt++) return 0; @@ -721,8 +708,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_SetupPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "SetupForPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -734,8 +721,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "TriggerPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (korg1212->playcnt++) return 0; @@ -743,8 +730,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) snd_korg1212_setCardState(korg1212, K1212_STATE_PLAYING); rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_TriggerPlay, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "TriggerPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -754,8 +741,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) /* spinlock already held */ static int snd_korg1212_StopPlay(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: StopPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "StopPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (--(korg1212->playcnt)) return 0; @@ -866,8 +853,9 @@ static int snd_korg1212_SetRate(struct snd_korg1212 *korg1212, int rate) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); return 0; } @@ -902,8 +890,8 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) u16 count; unsigned long flags; - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "WriteADCSensivity [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------------------------------- // initialize things. The local init bit is always set when writing to the @@ -1026,8 +1014,9 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_MonitorOn, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "WriteADCSensivity - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -1039,8 +1028,8 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) { int channel, rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is complete. [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is complete. [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------- // tell the card to boot @@ -1048,8 +1037,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_BootFromDSPPage4, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Boot from Page 4 - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Boot from Page 4 - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); msleep(DSP_BOOT_DELAY_IN_MS); // -------------------------------------------------------------------------------- @@ -1065,8 +1055,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Buffer Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Configure Buffer Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); udelay(INTERCOMMAND_DELAY); @@ -1079,8 +1070,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Misc Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Configure Misc Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); // -------------------------------------------------------------------------------- // Initialize the routing and volume tables, then update the card's state. @@ -1100,15 +1092,17 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); rc = snd_korg1212_TurnOnIdleMonitor(korg1212); snd_korg1212_setCardState(korg1212, K1212_STATE_READY); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Monitor On - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "K1212_DEBUG: Set Monitor On - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); snd_korg1212_setCardState(korg1212, K1212_STATE_DSP_COMPLETE); } @@ -1133,9 +1127,10 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) switch (doorbellValue) { case K1212_DB_DSPDownloadDone: - K1212_DEBUG_PRINTK("K1212_DEBUG: IRQ DNLD count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DNLD count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); if (korg1212->cardState == K1212_STATE_DSP_IN_PROCESS) { korg1212->dsp_is_loaded = 1; wake_up(&korg1212->wait); @@ -1146,10 +1141,11 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // an error occurred - stop the card // ------------------------------------------------------------------------ case K1212_DB_DMAERROR: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DMAE count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); - snd_printk(KERN_ERR "korg1212: DMA Error\n"); + dev_dbg(korg1212->card->dev, + "IRQ DMAE count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); + dev_err(korg1212->card->dev, "DMA Error\n"); korg1212->errorcnt++; korg1212->totalerrorcnt++; korg1212->sharedBufferPtr->cardCommand = 0; @@ -1161,16 +1157,19 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // the semaphore in case someone is waiting for this. // ------------------------------------------------------------------------ case K1212_DB_CARDSTOPPED: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ CSTP count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ CSTP count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); korg1212->sharedBufferPtr->cardCommand = 0; break; default: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", - korg1212->irqcount, doorbellValue, - korg1212->currentBuffer, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", + korg1212->irqcount, doorbellValue, + korg1212->currentBuffer, + stateName[korg1212->cardState]); if ((korg1212->cardState > K1212_STATE_SETUP) || korg1212->idleMonitorOn) { korg1212->currentBuffer++; @@ -1206,8 +1205,8 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is starting... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is starting... [%s]\n", + stateName[korg1212->cardState]); // --------------------------------------------------------------- // verify the state of the card before proceeding. @@ -1221,8 +1220,9 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) UpperWordSwap(korg1212->dma_dsp.addr), 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Start DSP Download RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Start DSP Download RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); korg1212->dsp_is_loaded = 0; wait_event_timeout(korg1212->wait, korg1212->dsp_is_loaded, HZ * CARD_BOOT_TIMEOUT); @@ -1281,20 +1281,20 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "K1212_DEBUG: snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", - dst, i); + dev_dbg(korg1212->card->dev, + "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", + dst, i); return -EFAULT; } -#endif memset((void*) dst + offset, 0, size); dst++; } @@ -1307,22 +1307,25 @@ static int snd_korg1212_copy_to(struct snd_korg1212 *korg1212, void __user *dst, struct KorgAudioFrame * src = korg1212->recordDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_to pos=%d offset=%d size=%d\n", - pos, offset, size); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to pos=%d offset=%d size=%d\n", + pos, offset, size); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) src < (void *) korg1212->recordDataBufsPtr || (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_to_user(dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } src++; @@ -1337,23 +1340,26 @@ static int snd_korg1212_copy_from(struct snd_korg1212 *korg1212, void __user *sr struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_from_user((void*) dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } dst++; @@ -1367,8 +1373,8 @@ static void snd_korg1212_free_pcm(struct snd_pcm *pcm) { struct snd_korg1212 *korg1212 = pcm->private_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_free_pcm [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_free_pcm [%s]\n", + stateName[korg1212->cardState]); korg1212->pcm = NULL; } @@ -1379,8 +1385,8 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1408,8 +1414,8 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1435,8 +1441,8 @@ static int snd_korg1212_playback_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_close [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_silence(korg1212, 0, K1212_MAX_SAMPLES, 0, korg1212->channels * 2); @@ -1457,8 +1463,8 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_close [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1475,14 +1481,18 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) static int snd_korg1212_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_ioctl: cmd=%d\n", cmd); + struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); + + dev_dbg(korg1212->card->dev, "snd_korg1212_ioctl: cmd=%d\n", cmd); if (cmd == SNDRV_PCM_IOCTL1_CHANNEL_INFO ) { struct snd_pcm_channel_info *info = arg; info->offset = 0; info->first = info->channel * 16; info->step = 256; - K1212_DEBUG_PRINTK("K1212_DEBUG: channel_info %d:, offset=%ld, first=%d, step=%d\n", info->channel, info->offset, info->first, info->step); + dev_dbg(korg1212->card->dev, + "channel_info %d:, offset=%ld, first=%d, step=%d\n", + info->channel, info->offset, info->first, info->step); return 0; } @@ -1498,8 +1508,8 @@ static int snd_korg1212_hw_params(struct snd_pcm_substream *substream, pid_t this_pid; pid_t other_pid; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_hw_params [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_hw_params [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1546,15 +1556,16 @@ static int snd_korg1212_prepare(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_prepare [%s]\n", + stateName[korg1212->cardState]); spin_lock_irq(&korg1212->lock); /* FIXME: we should wait for ack! */ if (korg1212->stop_pending_cnt > 0) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare - Stop is pending... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "snd_korg1212_prepare - Stop is pending... [%s]\n", + stateName[korg1212->cardState]); spin_unlock_irq(&korg1212->lock); return -EAGAIN; /* @@ -1579,8 +1590,8 @@ static int snd_korg1212_trigger(struct snd_pcm_substream *substream, struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_trigger [%s] cmd=%d\n", - stateName[korg1212->cardState], cmd); + dev_dbg(korg1212->card->dev, "snd_korg1212_trigger [%s] cmd=%d\n", + stateName[korg1212->cardState], cmd); spin_lock(&korg1212->lock); switch (cmd) { @@ -1621,8 +1632,9 @@ static snd_pcm_uframes_t snd_korg1212_playback_pointer(struct snd_pcm_substream pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1634,8 +1646,8 @@ static snd_pcm_uframes_t snd_korg1212_capture_pointer(struct snd_pcm_substream * pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1648,8 +1660,9 @@ static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_from(korg1212, src, pos, count, 0, korg1212->channels * 2); @@ -1662,8 +1675,8 @@ static int snd_korg1212_playback_silence(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_silence [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_silence [%s]\n", + stateName[korg1212->cardState]); return snd_korg1212_silence(korg1212, pos, count, 0, korg1212->channels * 2); } @@ -1676,8 +1689,9 @@ static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_capture_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_to(korg1212, dst, pos, count, 0, korg1212->channels * 2); } @@ -2138,7 +2152,6 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) static int snd_korg1212_dev_free(struct snd_device *device) { struct snd_korg1212 *korg1212 = device->device_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: Freeing device\n"); return snd_korg1212_free(korg1212); } @@ -2210,19 +2223,15 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, ioport_size = pci_resource_len(korg1212->pci, 1); iomem2_size = pci_resource_len(korg1212->pci, 2); - K1212_DEBUG_PRINTK("K1212_DEBUG: resources:\n" - " iomem = 0x%lx (%d)\n" - " ioport = 0x%lx (%d)\n" - " iomem = 0x%lx (%d)\n" - " [%s]\n", - korg1212->iomem, iomem_size, - korg1212->ioport, ioport_size, - korg1212->iomem2, iomem2_size, - stateName[korg1212->cardState]); + dev_dbg(card->dev, + "resources:\n\tiomem = 0x%lx (%d)\n\tioport = 0x%lx (%d)\n\tiomem = 0x%lx (%d)\n\t[%s]\n", + korg1212->iomem, iomem_size, korg1212->ioport, ioport_size, + korg1212->iomem2, iomem2_size, stateName[korg1212->cardState]); if ((korg1212->iobase = ioremap(korg1212->iomem, iomem_size)) == NULL) { - snd_printk(KERN_ERR "korg1212: unable to remap memory region 0x%lx-0x%lx\n", korg1212->iomem, - korg1212->iomem + iomem_size - 1); + dev_dbg(card->dev, + "unable to remap memory region 0x%lx-0x%lx\n", + korg1212->iomem, korg1212->iomem + iomem_size - 1); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2232,7 +2241,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, KBUILD_MODNAME, korg1212); if (err) { - snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq); + dev_dbg(card->dev, "unable to grab IRQ %d\n", + pci->irq); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2252,40 +2262,46 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->sensRegPtr = (u16 __iomem *) (korg1212->iobase + SENS_CONTROL_OFFSET); korg1212->idRegPtr = (u32 __iomem *) (korg1212->iobase + DEV_VEND_ID_OFFSET); - K1212_DEBUG_PRINTK("K1212_DEBUG: card registers:\n" - " Status register = 0x%p\n" - " OutDoorbell = 0x%p\n" - " InDoorbell = 0x%p\n" - " Mailbox0 = 0x%p\n" - " Mailbox1 = 0x%p\n" - " Mailbox2 = 0x%p\n" - " Mailbox3 = 0x%p\n" - " ControlReg = 0x%p\n" - " SensReg = 0x%p\n" - " IDReg = 0x%p\n" - " [%s]\n", - korg1212->statusRegPtr, - korg1212->outDoorbellPtr, - korg1212->inDoorbellPtr, - korg1212->mailbox0Ptr, - korg1212->mailbox1Ptr, - korg1212->mailbox2Ptr, - korg1212->mailbox3Ptr, - korg1212->controlRegPtr, - korg1212->sensRegPtr, - korg1212->idRegPtr, - stateName[korg1212->cardState]); + dev_dbg(card->dev, + "card registers:\n" + "\tStatus register = 0x%p\n" + "\tOutDoorbell = 0x%p\n" + "\tInDoorbell = 0x%p\n" + "\tMailbox0 = 0x%p\n" + "\tMailbox1 = 0x%p\n" + "\tMailbox2 = 0x%p\n" + "\tMailbox3 = 0x%p\n" + "\tControlReg = 0x%p\n" + "\tSensReg = 0x%p\n" + "\tIDReg = 0x%p\n" + "\t[%s]\n", + korg1212->statusRegPtr, + korg1212->outDoorbellPtr, + korg1212->inDoorbellPtr, + korg1212->mailbox0Ptr, + korg1212->mailbox1Ptr, + korg1212->mailbox2Ptr, + korg1212->mailbox3Ptr, + korg1212->controlRegPtr, + korg1212->sensRegPtr, + korg1212->idRegPtr, + stateName[korg1212->cardState]); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%Zd bytes)\n", sizeof(struct KorgSharedBuffer)); + dev_err(card->dev, + "can not allocate shared buffer memory (%Zd bytes)\n", + sizeof(struct KorgSharedBuffer)); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->sharedBufferPtr = (struct KorgSharedBuffer *)korg1212->dma_shared.area; korg1212->sharedBufferPhy = korg1212->dma_shared.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Shared Buffer Area = 0x%p (0x%08lx), %d bytes\n", korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, sizeof(struct KorgSharedBuffer)); + dev_dbg(card->dev, + "Shared Buffer Area = 0x%p (0x%08lx), %d bytes\n", + korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, + sizeof(struct KorgSharedBuffer)); #ifndef K1212_LARGEALLOC @@ -2293,27 +2309,35 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_play) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate play data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(card->dev, + "can not allocate play data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->playDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_play.area; korg1212->PlayDataPhy = korg1212->dma_play.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Play Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->playDataBufsPtr, korg1212->PlayDataPhy, korg1212->DataBufsSize); + dev_dbg(card->dev, + "Play Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->playDataBufsPtr, korg1212->PlayDataPhy, + korg1212->DataBufsSize); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_rec) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate record data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(card->dev, + "can not allocate record data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->recordDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_rec.area; korg1212->RecDataPhy = korg1212->dma_rec.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Record Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->recordDataBufsPtr, korg1212->RecDataPhy, korg1212->DataBufsSize); + dev_dbg(card->dev, + "Record Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->recordDataBufsPtr, korg1212->RecDataPhy, + korg1212->DataBufsSize); #else // K1212_LARGEALLOC @@ -2334,22 +2358,25 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev); if (err < 0) { release_firmware(dsp_code); - snd_printk(KERN_ERR "firmware not available\n"); + dev_err(card->dev, "firmware not available\n"); snd_korg1212_free(korg1212); return err; } if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), dsp_code->size, &korg1212->dma_dsp) < 0) { - snd_printk(KERN_ERR "korg1212: cannot allocate dsp code memory (%zd bytes)\n", dsp_code->size); + dev_err(card->dev, + "cannot allocate dsp code memory (%zd bytes)\n", + dsp_code->size); snd_korg1212_free(korg1212); release_firmware(dsp_code); return -ENOMEM; } - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", - korg1212->dma_dsp.area, korg1212->dma_dsp.addr, dsp_code->size, - stateName[korg1212->cardState]); + dev_dbg(card->dev, + "DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", + korg1212->dma_dsp.area, korg1212->dma_dsp.addr, dsp_code->size, + stateName[korg1212->cardState]); memcpy(korg1212->dma_dsp.area, dsp_code->data, dsp_code->size); @@ -2358,7 +2385,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_RebootCard, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Reboot Card - RC = %d [%s]\n", rc, stateName[korg1212->cardState]); + dev_dbg(card->dev, "Reboot Card - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, korg1212, &ops)) < 0) { snd_korg1212_free(korg1212); @@ -2372,18 +2400,22 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_korg1212_downloadDSPCode(korg1212)) return -EBUSY; - K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " - "PlayDataPhy = %08x L[%08x]\n" - "korg1212: RecDataPhy = %08x L[%08x], " - "VolumeTablePhy = %08x L[%08x]\n" - "korg1212: RoutingTablePhy = %08x L[%08x], " - "AdatTimeCodePhy = %08x L[%08x]\n", - (int)korg1212->dma_dsp.addr, UpperWordSwap(korg1212->dma_dsp.addr), - korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy), - korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), - korg1212->VolumeTablePhy, LowerWordSwap(korg1212->VolumeTablePhy), - korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), - korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); + dev_dbg(card->dev, + "dspMemPhy = %08x U[%08x], PlayDataPhy = %08x L[%08x]\n", + (int)korg1212->dma_dsp.addr, + UpperWordSwap(korg1212->dma_dsp.addr), + korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy)); + dev_dbg(card->dev, + "RecDataPhy = %08x L[%08x], VolumeTablePhy = %08x L[%08x]\n", + korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), + korg1212->VolumeTablePhy, + LowerWordSwap(korg1212->VolumeTablePhy)); + dev_dbg(card->dev, + "RoutingTablePhy = %08x L[%08x], AdatTimeCodePhy = %08x L[%08x]\n", + korg1212->RoutingTablePhy, + LowerWordSwap(korg1212->RoutingTablePhy), + korg1212->AdatTimeCodePhy, + LowerWordSwap(korg1212->AdatTimeCodePhy)); if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) return err; @@ -2446,7 +2478,7 @@ snd_korg1212_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, korg1212->iomem, korg1212->irq); - K1212_DEBUG_PRINTK("K1212_DEBUG: %s\n", card->longname); + dev_dbg(card->dev, "%s\n", card->longname); if ((err = snd_card_register(card)) < 0) { snd_card_free(card); -- 1.8.1.2

2 4

[alsa-devel] [PATCH v3 0/6] ASoC: Intel: ACPI support and machines
by Vinod Koul 10 Nov '14

10 Nov '14

This is third rev of patch series which fixes the modules split as discussed in ML. The core part is now module along with PCI and ACPI one. Added fix reported by Dan as well. Also added the BYTCR machine driver and MERR machine driver Updates in v3: - Fix export fo few more symbols - fix cleanup of probe and setup code - add MERR machine code Subhransu S. Prusty (1): ASoC: Intel: add BYTCR machine driver with RT5640 Vinod Koul (5): ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld- add ACPI module ASoC: Intel: Add Merrifield machine driver sound/soc/intel/Kconfig | 37 +++- sound/soc/intel/Makefile | 4 + sound/soc/intel/bytcr_dpcm_rt5640.c | 258 ++++++++++++++++++++++ sound/soc/intel/merr_dpcm_wm8958.c | 416 +++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.c | 2 +- sound/soc/intel/sst/Makefile | 8 +- sound/soc/intel/sst/sst.c | 254 +++++---------------- sound/soc/intel/sst/sst.h | 9 +- sound/soc/intel/sst/sst_acpi.c | 377 +++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pci.c | 209 ++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 3 + 11 files changed, 1378 insertions(+), 199 deletions(-) create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/merr_dpcm_wm8958.c create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_pci.c

2 20

[alsa-devel] [PATCH 0/3 v4] Digidesign Mbox 1 enhancements
by Damien Zammit 10 Nov '14

10 Nov '14

This is fourth revision based on all reviews/tweaks so far. Damien Zammit (3): ALSA: usb-audio: Move mutex define to more general location ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/card.c | 1 - sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 144 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 ++++++++++---- sound/usb/usbaudio.h | 2 + 5 files changed, 188 insertions(+), 11 deletions(-) -- 1.9.1

3 7

[alsa-devel] [PATCH v4 0/4] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 10 Nov '14

10 Nov '14

This patch set adds multi-CODEC support to the simple card. v4: (Mark Brown) - document the new core function - calculation of the number of codecs moved to the core function - split the patch about simple-card.c - less code changes - remove the cast of pointers - less changes in the DT binding example 2 v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (4): ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: simple-card: add multi-CODECs in DT ASoC: simple-card: Remove useless check .../devicetree/bindings/sound/simple-card.txt | 20 ++-- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 87 ++++++++-------- sound/soc/soc-core.c | 113 ++++++++++++++++++--- 4 files changed, 154 insertions(+), 69 deletions(-) -- 2.1.3

2 7

[alsa-devel] [PATCH 0/2 v3] Digidesign Mbox 1 enhancements
by Damien Zammit 10 Nov '14

10 Nov '14

This is the third patchset to provide mixer control and duplex mode. Was rebased on top of 85a8181329a and tested on the hardware. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 142 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 ++++++++++---- 3 files changed, 184 insertions(+), 10 deletions(-) -- 1.9.1

2 3

[alsa-devel] [PATCH 0/2] Digidesign Mbox 1 duplex mode and mixer
by Damien Zammit 10 Nov '14

10 Nov '14

Takashi asked me to rewrite the patches and send them without attachments. I hope this email series works and they can be reviewed. These two patches provide the following for Digidesign Mbox 1: o Mixer control for clock source selection: Internal/SPDIF o Duplex mode (playback and capture) Damien Zammit (2): snd-usb-audio: Add mixer control for Digidesign Mbox 1 clock source snd-usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/card.h | 5 ++ sound/usb/mixer_maps.c | 9 +++ sound/usb/mixer_quirks.c | 177 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 73 ++++++++++++------- sound/usb/quirks.c | 30 ++++++++ sound/usb/usbaudio.h | 1 + 6 files changed, 271 insertions(+), 24 deletions(-) -- 1.9.1

3 14

[alsa-devel] [PATCH v3 0/2] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 09 Nov '14

09 Nov '14

This patch set adds multi-CODEC support to the simple card. v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (2): ASoC: core: add multi-codec support in DT ASoC: simple-card: add multi-CODECs in DT .../devicetree/bindings/sound/simple-card.txt | 28 ++--- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 135 +++++++++++++-------- sound/soc/soc-core.c | 75 +++++++++--- 4 files changed, 157 insertions(+), 84 deletions(-) -- 2.1.1

3 8

[alsa-devel] NXP Semiconductors TFA9879 Amplifier Driver
by Peter Rosin 09 Nov '14

09 Nov '14

Hi! Sorry for not sending this from my axentia.se account, but I tend to get high spam-scores from there when I use git send-email. This is a new driver, and it's pretty minimalistic with support for only a few basic controls. However, it is usable and I'd be happy to see it included. I don't know if it's presumptious to add myself as a maintainer, but checkpatch complained... Are the alsa control names wisely chosen? Please review! Cheers, Peter

5 15

[alsa-devel] [PATCH 1/2] ASoC: Forward calls to snd_soc_cache_sync() to regcache_sync()
by Lars-Peter Clausen 09 Nov '14

09 Nov '14

For convenience for drivers that do not want to keep their own pointer to regmap struct around forward calls to snd_soc_cache_sync() to regcache_sync() if the driver is using regmap. This is similar to what we do for snd_soc_read()/snd_soc_write(). This patch also fixes drivers which already have been converted to regmap, but still use snd_soc_cache_sync() for trying to the sync the cache. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Unfortunately there are a few drivers that have already been converted to regmap, but still use snd_soc_cache_sync(). snd_soc_cache_sync() currently is a noop for those drivers. Some of these conversions were done as early as 3.4 and while this patch applies cleanly to earlier kernel versions, due to the restructuring of the ASoC core it will only work on 3.16+. Hence no Fixes tag. --- sound/soc/soc-cache.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a9f82b5..6dab817 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -187,6 +187,9 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) const char *name = "flat"; int ret; + if (codec->component.regmap) + return regcache_sync(codec->component.regmap); + if (!codec->cache_sync) return 0; -- 1.8.0

2 2

[alsa-devel] [PATCH 0/2 v2] Digidesign Mbox 1 clock source and duplex mode
by damien.zammit＠gmail.com 09 Nov '14

09 Nov '14

From: Damien Zammit <damien(a)zamaudio.com> This is a cleaned up patchset version 2. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/card.h | 5 ++ sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 133 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 73 +++++++++++++++++--------- sound/usb/quirks.c | 30 +++++++++++ sound/usb/usbaudio.h | 1 + 6 files changed, 227 insertions(+), 24 deletions(-) -- 1.9.1

1 2

[alsa-devel] [PATCH 0/2] ASoC: More AC'97 cleanups
by Lars-Peter Clausen 08 Nov '14

08 Nov '14

More AC'97 cleanups in preparation for the conversion to regmap. * The mioa701_wm9713 directly accesses the read/write callback of the CODEC driver which will break when converting the driver to regmap. * Re-worked version of the wm9712/wm9713 virtual register removal, this time using custom put/get handlers to keep the control names. - Lars Lars-Peter Clausen (2): ASoC: mioa701_wm9713: Don't opencode CODEC register access ASoC: wm9712/wm9713: Replace virtual registers with custom put/get callbacks sound/soc/codecs/wm9712.c | 163 ++++++++++++++++++++++++++--------------- sound/soc/codecs/wm9713.c | 153 ++++++++++++++++++++++---------------- sound/soc/pxa/mioa701_wm9713.c | 7 +- 3 files changed, 196 insertions(+), 127 deletions(-) -- 1.8.0

3 6

[alsa-devel] newbie at helping alsa
by John Houston 08 Nov '14

08 Nov '14

I am a retired engineer with some years of experience. I would like to help with the ALC892 portion of the project. I have an ALC892 in my computer and would like to use it. I also have an application for it.

1 0

[alsa-devel] [PATCH] ALSA: pcm: Add snd_pcm_stop_xrun() helper
by Takashi Iwai 07 Nov '14

07 Nov '14

Add a new helper function snd_pcm_stop_xrun() to the standard sequnce lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the existing open codes with this helper. The function checks the PCM running state to prevent setting the wrong state, too, for more safety. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/media/pci/saa7134/saa7134-alsa.c | 4 +--- include/sound/pcm.h | 1 + sound/arm/pxa2xx-pcm-lib.c | 4 +--- sound/core/pcm_native.c | 23 +++++++++++++++++++++++ sound/firewire/amdtp.c | 8 ++------ sound/firewire/isight.c | 10 ++-------- sound/pci/asihpi/asihpi.c | 5 +---- sound/pci/atiixp.c | 4 +--- sound/pci/atiixp_modem.c | 4 +--- sound/soc/atmel/atmel-pcm-dma.c | 4 +--- sound/soc/fsl/fsl_dma.c | 9 +-------- sound/usb/6fire/pcm.c | 17 ++++------------- sound/usb/endpoint.c | 4 +--- sound/usb/misc/ua101.c | 18 ++++-------------- sound/usb/usx2y/usbusx2yaudio.c | 9 ++------- 15 files changed, 46 insertions(+), 78 deletions(-) diff --git a/drivers/media/pci/saa7134/saa7134-alsa.c b/drivers/media/pci/saa7134/saa7134-alsa.c index 40569894c1c9..ac3cd74e824e 100644 --- a/drivers/media/pci/saa7134/saa7134-alsa.c +++ b/drivers/media/pci/saa7134/saa7134-alsa.c @@ -173,9 +173,7 @@ static void saa7134_irq_alsa_done(struct saa7134_dev *dev, dprintk("irq: overrun [full=%d/%d] - Blocks in %d\n",dev->dmasound.read_count, dev->dmasound.bufsize, dev->dmasound.blocks); spin_unlock(&dev->slock); - snd_pcm_stream_lock(dev->dmasound.substream); - snd_pcm_stop(dev->dmasound.substream,SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dev->dmasound.substream); + snd_pcm_stop_xrun(dev->dmasound.substream); return; } diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0b8daeed0a33..40289ec2451c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -506,6 +506,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream, int snd_pcm_start(struct snd_pcm_substream *substream); int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); +int snd_pcm_stop_xrun(struct snd_pcm_substream *substream); #ifdef CONFIG_PM int snd_pcm_suspend(struct snd_pcm_substream *substream); int snd_pcm_suspend_all(struct snd_pcm *pcm); diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index a61d7a9a995e..01f8fdc42b1b 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -200,9 +200,7 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) } else { printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", dma_ch, dcsr); - snd_pcm_stream_lock(substream); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(substream); + snd_pcm_stop_xrun(substream); } } EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index dfb5031969f8..a3d122109704 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1098,6 +1098,29 @@ int snd_pcm_drain_done(struct snd_pcm_substream *substream) SNDRV_PCM_STATE_SETUP); } +/** + * snd_pcm_stop_xrun - stop the running streams as XRUN + * @substream: the PCM substream instance + * @state: PCM state after stopping the stream + * + * This stops the given running substream (and all linked substreams) as XRUN. + * Unlike snd_pcm_stop(), this function takes the substream lock by itself. + * + * Return: Zero if successful, or a negative error code. + */ +int snd_pcm_stop_xrun(struct snd_pcm_substream *substream) +{ + unsigned long flags; + int ret = 0; + + snd_pcm_stream_lock_irqsave(substream, flags); + if (snd_pcm_running(substream)) + ret = snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(substream, flags); + return ret; +} +EXPORT_SYMBOL_GPL(snd_pcm_stop_xrun); + /* * pause callbacks */ diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 95fc2eaf11dc..3badc70124ab 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -1006,11 +1006,7 @@ void amdtp_stream_pcm_abort(struct amdtp_stream *s) struct snd_pcm_substream *pcm; pcm = ACCESS_ONCE(s->pcm); - if (pcm) { - snd_pcm_stream_lock_irq(pcm); - if (snd_pcm_running(pcm)) - snd_pcm_stop(pcm, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irq(pcm); - } + if (pcm) + snd_pcm_stop_xrun(pcm); } EXPORT_SYMBOL(amdtp_stream_pcm_abort); diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 7ac94439e758..48d6dca471c6 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -131,14 +131,8 @@ static void isight_samples(struct isight *isight, static void isight_pcm_abort(struct isight *isight) { - unsigned long flags; - - if (ACCESS_ONCE(isight->pcm_active)) { - snd_pcm_stream_lock_irqsave(isight->pcm, flags); - if (snd_pcm_running(isight->pcm)) - snd_pcm_stop(isight->pcm, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(isight->pcm, flags); - } + if (ACCESS_ONCE(isight->pcm_active)) + snd_pcm_stop_xrun(isight->pcm); } static void isight_dropped_samples(struct isight *isight, unsigned int total) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index ac66b3228a34..ff9f9f1c0391 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -769,10 +769,7 @@ static void snd_card_asihpi_timer_function(unsigned long data) s->number); ds->drained_count++; if (ds->drained_count > 20) { - unsigned long flags; - snd_pcm_stream_lock_irqsave(s, flags); - snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(s, flags); + snd_pcm_stop_xrun(s); continue; } } else { diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7895c5d300c7..9c1c4452a8ee 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -688,9 +688,7 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) if (! dma->substream || ! dma->running) return; dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type); - snd_pcm_stream_lock(dma->substream); - snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dma->substream); + snd_pcm_stop_xrun(dma->substream); } /* diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 3c3241309a30..b2f63e0727de 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -638,9 +638,7 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, if (! dma->substream || ! dma->running) return; dev_dbg(chip->card->dev, "XRUN detected (DMA %d)\n", dma->ops->type); - snd_pcm_stream_lock(dma->substream); - snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(dma->substream); + snd_pcm_stop_xrun(dma->substream); } /* diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b79a2a864513..33fb3bb133df 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -80,9 +80,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, /* stop RX and capture: will be enabled again at restart */ ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable); - snd_pcm_stream_lock(substream); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock(substream); + snd_pcm_stop_xrun(substream); /* now drain RHR and read status to remove xrun condition */ ssc_readx(prtd->ssc->regs, SSC_RHR); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index a609aafc994d..b2b108805b24 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -151,14 +151,7 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { */ static void fsl_dma_abort_stream(struct snd_pcm_substream *substream) { - unsigned long flags; - - snd_pcm_stream_lock_irqsave(substream, flags); - - if (snd_pcm_running(substream)) - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - - snd_pcm_stream_unlock_irqrestore(substream, flags); + snd_pcm_stop_xrun(substream); } /** diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index ba40489b2de4..36f4115eb1cd 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -679,25 +679,16 @@ int usb6fire_pcm_init(struct sfire_chip *chip) void usb6fire_pcm_abort(struct sfire_chip *chip) { struct pcm_runtime *rt = chip->pcm; - unsigned long flags; int i; if (rt) { rt->panic = true; - if (rt->playback.instance) { - snd_pcm_stream_lock_irqsave(rt->playback.instance, flags); - snd_pcm_stop(rt->playback.instance, - SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags); - } + if (rt->playback.instance) + snd_pcm_stop_xrun(rt->playback.instance); - if (rt->capture.instance) { - snd_pcm_stream_lock_irqsave(rt->capture.instance, flags); - snd_pcm_stop(rt->capture.instance, - SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags); - } + if (rt->capture.instance) + snd_pcm_stop_xrun(rt->capture.instance); for (i = 0; i < PCM_N_URBS; i++) { usb_poison_urb(&rt->in_urbs[i].instance); diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a4679913b0aa..03b074419964 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -391,9 +391,7 @@ static void snd_complete_urb(struct urb *urb) usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err); if (ep->data_subs && ep->data_subs->pcm_substream) { substream = ep->data_subs->pcm_substream; - snd_pcm_stream_lock_irqsave(substream, flags); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(substream, flags); + snd_pcm_stop_xrun(substream); } exit_clear: diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index a1bab149df4d..9581089c28c5 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -613,24 +613,14 @@ static int start_usb_playback(struct ua101 *ua) static void abort_alsa_capture(struct ua101 *ua) { - unsigned long flags; - - if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) { - snd_pcm_stream_lock_irqsave(ua->capture.substream, flags); - snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags); - } + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + snd_pcm_stop_xrun(ua->capture.substream); } static void abort_alsa_playback(struct ua101 *ua) { - unsigned long flags; - - if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) { - snd_pcm_stream_lock_irqsave(ua->playback.substream, flags); - snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags); - } + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + snd_pcm_stop_xrun(ua->playback.substream); } static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index a63330dd1407..61d5dc2a3421 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -272,13 +272,8 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) for (s = 0; s < 4; s++) { struct snd_usX2Y_substream *subs = usX2Y->subs[s]; if (subs) { - if (atomic_read(&subs->state) >= state_PRERUNNING) { - unsigned long flags; - - snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags); - snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags); - } + if (atomic_read(&subs->state) >= state_PRERUNNING) + snd_pcm_stop_xrun(subs->pcm_substream); for (u = 0; u < NRURBS; u++) { struct urb *urb = subs->urb[u]; if (NULL != urb) -- 2.1.3

1 0

[alsa-devel] [PATCH 1/4] ASoC:rt5670:Add runtime PM support
by bardliao＠realtek.com 07 Nov '14

07 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds runtime PM support on rt5670 codec. And will switch sysclk to internal clock during runtime suspend. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index da4b689..97359b3 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,6 +14,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/firmware.h> @@ -2666,6 +2667,13 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, /*Give sysclk a default value*/ rt5670->sysclk = 24576000; + /* The set initial power status to active, to be consist with ACPI power + * state D0. The codec will be suspended after probe and runtime status + * change will trigger ACPI D3 method for power saving. + */ + pm_runtime_set_active(&i2c->dev); + pm_runtime_enable(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) @@ -2673,20 +2681,61 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; } +#ifdef CONFIG_PM_RUNTIME +static void rt5670_use_internal_clk(struct rt5670_priv *rt5670) +{ + struct snd_soc_codec *codec = rt5670->codec; + + if (!codec || + (0 == rt5670->sysclk && RT5670_SCLK_S_RCCLK == rt5670->sysclk_src)) + return; + + snd_soc_update_bits(codec, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); + + rt5670->sysclk = 0; + rt5670->sysclk_src = RT5670_SCLK_S_RCCLK; +} + +static int rt5670_runtime_suspend(struct device *dev) +{ + struct rt5670_priv *rt5670 = dev_get_drvdata(dev); + + rt5670_use_internal_clk(rt5670); + return 0; +} + +static int rt5670_runtime_resume(struct device *dev) +{ + return 0; +} + +static const struct dev_pm_ops rt5670_pm = { + SET_RUNTIME_PM_OPS(rt5670_runtime_suspend, rt5670_runtime_resume, NULL) +}; +#define RT5670_PM_OPS rt5670_pm +#else +#define RT5670_PM_OPS NULL +#endif /* CONFIG_PM_RUNTIME */ + static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", .owner = THIS_MODULE, + .pm = &RT5670_PM_OPS, }, .probe = rt5670_i2c_probe, .remove = rt5670_i2c_remove, -- 1.8.1.1.439.g50a6b54

3 10

[alsa-devel] [PATCH] ALSA: snd_ctl_activate_id(): Fix index look-up
by Lars-Peter Clausen 07 Nov '14

07 Nov '14

We want to know the offset for the id that was passed to the function, not the offset of the first id of the control (which is always 0). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- The function seems currently unused, so no need to add this for stable --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 99aa3aa..bb96a46 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -572,7 +572,7 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, ret = -ENOENT; goto unlock; } - index_offset = snd_ctl_get_ioff(kctl, &kctl->id); + index_offset = snd_ctl_get_ioff(kctl, id); vd = &kctl->vd[index_offset]; ret = 0; if (active) { -- 1.8.0

2 1

[alsa-devel] [PATCH 0/2] add support for exynos7 I2S controller
by Padmavathi Venna 07 Nov '14

07 Nov '14

I2S on exynos7 includes the below changes - has got support for lower root clock sampling frequencies like 64, 128, 96, 192fs. I2S on previous SoCs supports only 256,512,384 and 768. - supports 7.1CH TDM mode for recording. exynos5 has only 7.1CH TDM mode for playback. - secondary dai works with only external DMA. on previous SoCs secondary dai can work with either both or only internal DMA. - I2S1 controller upgraded to 5.1CH with slighlty modified bit offsets for supporting lower RFS and more no.of BFS values. As internal dma is not mandatory on all SoCs this patchset introduced a quirk for the same and added a structure to hold all the modified bit offsets across all SoCs. This patchset is made based on linux-next master branch and tested with exynos7 base support from the below link. http://www.spinics.net/lists/linux-samsung-soc/msg37047.html Padmavathi Venna (2): ASoC: Samsung: Add quirk for internal DMA ASoC: samsung: add support for exynos7 I2S controller .../devicetree/bindings/sound/samsung-i2s.txt | 15 +- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/i2s-regs.h | 10 +- sound/soc/samsung/i2s.c | 228 ++++++++++++++------ 5 files changed, 180 insertions(+), 76 deletions(-) -- 1.7.4.4

2 4

[alsa-devel] Serial MIDI driver for PL011 - as found on BCM2835 hardware
by Kim Taylor 07 Nov '14

07 Nov '14

Hi, I've written a driver for raw MIDI using the PL011. It is obviously based on the snd-serial-u16550 module, with a couple of improvements: - I've implemented a drain() callback. - I've also implemented a half duplex mode based on a timer callback which was necessary for reliable communication with my eMagic Unitor 8 hardware. (Testing with a serial link to other hardware shows that the full duplex mode is also working.) It can be found here: https://github.com/kmtaylor/rpi_patches/blob/master/serial-pl011.c If you think it should be included in the ALSA repository, please let me know what to do next. Regards, Kim

4 6

[alsa-devel] [PATCH 0/6] ASoC/ARM: move Marvell Armada 370 DB to simple-audio-card
by Thomas Petazzoni 07 Nov '14

07 Nov '14

Hello, This set of patches moves the Marvell Armada 370 DB audio complex description from a custom audio machine driver to the simple-audio-card DT binding. The first three patches are meant to go through the ASoC tree, the last three patches are meant to go through the mvebu maintainers tree. Quick description of the patches: - First patch fixes a small issue in the cs42l51 driver following its split in two parts. - Second patch makes the cs42l51 user-selectable by adding a prompt to its Kconfig option. - Third patch gets rid of the machine driver. - Fourth patch removes some now incorrect pin-muxing selection in the Armada 370 DB platform. - Fifth patch actually switches to use the simple-audio-card DT binding on Armada 370 DB. - Sixth patch updates mvebu_v7_defconfig to select the relevant options to get audio to work. Note: obviously, after PATCH 3 is applied, the audio will no longer work until PATCH 5 is applied. But since those patches will anyway go through different trees, it is a bit inevitable. And it is not dramatically important, since audio support on Armada 370 DB is already broken since the split of the CS42L51 driver. Thanks, Thomas Thomas Petazzoni (6): ASoC: cs42l51: re-add of_match_table pointer ASoC: cs42l51: make driver user-selectable ASoC: kirkwood: get rid of armada-370-db driver ARM: mvebu: remove conflicting muxing on Armada 370 DB ARM: mvebu: use simple-card DT binding for audio on Armada 370 DB ARM: mvebu: update mvebu_v7_defconfig for Armada 370 audio arch/arm/boot/dts/armada-370-db.dts | 80 +++++++++++-------- arch/arm/boot/dts/armada-370.dtsi | 1 + arch/arm/configs/mvebu_v7_defconfig | 4 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/cs42l51-i2c.c | 7 ++ sound/soc/codecs/cs42l51.c | 5 -- sound/soc/kirkwood/Kconfig | 13 +--- sound/soc/kirkwood/Makefile | 4 - sound/soc/kirkwood/armada-370-db.c | 148 ------------------------------------ 9 files changed, 63 insertions(+), 201 deletions(-) delete mode 100644 sound/soc/kirkwood/armada-370-db.c -- 2.0.0

5 17

[alsa-devel] [PATCH 0/8] ASoC: rsnd: add Volume Ramp support
by Kuninori Morimoto 06 Nov '14

06 Nov '14

Hi Mark These patches add Volume Ramp support on Renesas R-Car sound driver. Kuninori Morimoto (8): ASoC: rsnd: tidyup debug information when read/write ASoC: rsnd: tidyup RSND_DVC_VOLUME_NUM to RSND_DVC_CHANNELS ASoC: rsnd: add struct rsnd_dvc_cfg and control DVC settings ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: Add Volume Ramp support sound/soc/sh/rcar/dvc.c | 155 ++++++++++++++++++++++++++++++++++------------ sound/soc/sh/rcar/gen.c | 30 ++++++--- sound/soc/sh/rcar/rsnd.h | 6 ++ 3 files changed, 145 insertions(+), 46 deletions(-) Best regards --- Kuninori Morimoto

2 24

[alsa-devel] [PATCH] ASoC: Intel: mrfld - remove non static definition
by Vinod Koul 06 Nov '14

06 Nov '14

sst_save_shim64() is defined as static in code but header is non static. Since this is not used other than file where defined remove non static definition Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/sst/sst.h | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 3ee555e..2dcbf47 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -508,8 +508,6 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, size_t mbox_data_len, const void *mbox_data, void **data, bool large, bool fill_dsp, bool sync, bool response); -void sst_save_shim64(struct intel_sst_drv *ctx, void __iomem *shim, - struct sst_shim_regs64 *shim_regs); void sst_process_pending_msg(struct work_struct *work); int sst_assign_pvt_id(struct intel_sst_drv *sst_drv_ctx); void sst_init_stream(struct stream_info *stream, -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: ssm4567: Add DAC high-pass-filter control
by Lars-Peter Clausen 06 Nov '14

06 Nov '14

Add a switch which can be used to enable/disable the DAC high-pass-filter. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm4567.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index 4b5c17f..e1e33d8 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -145,6 +145,8 @@ static const struct snd_kcontrol_new ssm4567_snd_controls[] = { SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0, 0xff, 1, ssm4567_vol_tlv), SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0), + SOC_SINGLE("DAC High Pass Filter Switch", SSM4567_REG_DAC_CTRL, + 5, 1, 0), }; static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { -- 1.8.0

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: Trigger PCM XRUN at XRUN
by Takashi Iwai 06 Nov '14

06 Nov '14

The usb-audio driver detects XRUN at its complete callback, but the actual code to trigger PCM XRUN is commented out because it caused deadlock in the past. This patch revives the PCM trigger properly. It resulted in more than just enabling snd_pcm_stop(), but it had to deduce the PCM substream with proper NULL checks and holds the stream lock around the call. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 167d0c1643e1..a4679913b0aa 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -348,6 +348,8 @@ static void snd_complete_urb(struct urb *urb) { struct snd_urb_ctx *ctx = urb->context; struct snd_usb_endpoint *ep = ctx->ep; + struct snd_pcm_substream *substream; + unsigned long flags; int err; if (unlikely(urb->status == -ENOENT || /* unlinked */ @@ -364,8 +366,6 @@ static void snd_complete_urb(struct urb *urb) goto exit_clear; if (snd_usb_endpoint_implicit_feedback_sink(ep)) { - unsigned long flags; - spin_lock_irqsave(&ep->lock, flags); list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); spin_unlock_irqrestore(&ep->lock, flags); @@ -389,7 +389,12 @@ static void snd_complete_urb(struct urb *urb) return; usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err); - //snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + if (ep->data_subs && ep->data_subs->pcm_substream) { + substream = ep->data_subs->pcm_substream; + snd_pcm_stream_lock_irqsave(substream, flags); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(substream, flags); + } exit_clear: clear_bit(ctx->index, &ep->active_mask); -- 2.1.3

1 0

[alsa-devel] [PATCH] ASoC:rt5645:Add JD function support
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> rt5645 codec support jack detection function. The patch will set related registers if JD function is used. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5645.h | 3 +++ sound/soc/codecs/rt5645.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt5645.h | 5 +++++ 3 files changed, 28 insertions(+) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index a535271..937f421 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -23,6 +23,9 @@ struct rt5645_platform_data { unsigned int hp_det_gpio; bool gpio_hp_det_active_high; + + /* true if codec's jd function is used */ + bool en_jd_func; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 0e92e63..1c099ee 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2205,6 +2205,13 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); + /* for JD function */ + if (rt5645->pdata.en_jd_func) { + snd_soc_dapm_force_enable_pin(&codec->dapm, "JD Power"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); + snd_soc_dapm_sync(&codec->dapm); + } + return 0; } @@ -2438,6 +2445,19 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } + if (rt5645->pdata.en_jd_func) { + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU, + RT5645_IRQ_CLK_GATE_CTRL | RT5645_MICINDET_MANU); + regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1, + RT5645_CBJ_BST1_EN, RT5645_CBJ_BST1_EN); + regmap_update_bits(rt5645->regmap, RT5645_JD_CTRL3, + RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL, + RT5645_JD_CBJ_EN | RT5645_JD_CBJ_POL); + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT); + } + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 5ec2520..82f681b 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1348,6 +1348,8 @@ #define RT5645_PWR_CLK25M_SFT 4 #define RT5645_PWR_CLK25M_PD (0x0 << 4) #define RT5645_PWR_CLK25M_PU (0x1 << 4) +#define RT5645_IRQ_CLK_MCLK (0x0 << 3) +#define RT5645_IRQ_CLK_INT (0x1 << 3) /* VAD Control 4 (0x9d) */ #define RT5645_VAD_SEL_MASK (0x3 << 8) @@ -2116,6 +2118,9 @@ enum { #define RT5645_RXDP2_SEL_ADC (0x1 << 3) #define RT5645_RXDP2_SEL_SFT (3) +/* General Control3 (0xfc) */ +#define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) +#define RT5645_MICINDET_MANU (0x1 << 7) /* Vendor ID (0xfd) */ #define RT5645_VER_C 0x2 -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] [PATCH] ALSA: pcm: Update the state properly before notification
by Takashi Iwai 06 Nov '14

06 Nov '14

Some state changes (e.g. snd_pcm_stop()) sets the runtime state after calling snd_timer_notify(). This is basically racy, since the notification may wakes up the user even before the state change. Although the possibility is low, we should set the state before the notifications. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ca224fa2a33a..dfb5031969f8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1052,10 +1052,10 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); + runtime->status->state = state; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, &runtime->trigger_tstamp); - runtime->status->state = state; } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1204,11 +1204,11 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->status->suspended_state = runtime->status->state; + runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, &runtime->trigger_tstamp); - runtime->status->suspended_state = runtime->status->state; - runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1311,10 +1311,10 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); + runtime->status->state = runtime->status->suspended_state; if (substream->timer) snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, &runtime->trigger_tstamp); - runtime->status->state = runtime->status->suspended_state; } static struct action_ops snd_pcm_action_resume = { -- 2.1.3

1 0

[alsa-devel] [PATCH alsa-lib] pcm: dmix: Don't ignore SND_TIMER_EVENT_MSTOP
by Takashi Iwai 06 Nov '14

06 Nov '14

When a slave PCM gets an error like XRUN, it stops and notifies with SND_TIMER_EVENT_MSTOP event. But the current code filters out this type and eventually hang due to the empty timer queue. The fix is to just add this event type to the filter bit mask. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 8e37bcba5e19..195fddf06cda 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1148,6 +1148,7 @@ int snd_pcm_direct_initialize_poll_fd(snd_pcm_direct_t *dmix) dmix->timer_events = (1<<SND_TIMER_EVENT_MSUSPEND) | (1<<SND_TIMER_EVENT_MRESUME) | + (1<<SND_TIMER_EVENT_MSTOP) | (1<<SND_TIMER_EVENT_STOP); /* -- 2.1.3

1 0

[alsa-devel] [PATCH 1/2] ASoC:rt286: set combo jack by dmi
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch enables combo jack configuration according to dmi. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * rename intel_platform to force_combo_jack_table --- --- sound/soc/codecs/rt286.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 97daa80..d4acb64 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1205,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +static struct dmi_system_id force_combo_jack_table[] __initdata = { + { + .ident = "Intel Wilson Beach", + .matches = { + DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1240,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + if (dmi_check_system(force_combo_jack_table)) + rt286->pdata.cbj_en = true; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Remove '.owner' field of driver
by Manish Badarkhe 06 Nov '14

06 Nov '14

Remove '.owner' field of driver structure as it is going to be filled via "module_i2c_driver" call. Signed-off-by: Manish Badarkhe <manishvb(a)ti.com> --- :100644 100644 0307909... 191b423... M sound/soc/codecs/tlv320aic31xx.c sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0307909..191b423 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1303,7 +1303,6 @@ MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, -- 1.7.9.5

1 0

Re: [alsa-devel] Restore hidden register for default value
by Kailang 06 Nov '14

06 Nov '14

Hi Takashi, I mail it again. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Friday, October 31, 2014 3:12 PM > To: 'Takashi Iwai' > Cc: (alsa-devel(a)alsa-project.org) > Subject: RE: Restore hidden register for default value > > Hi Takashi, > > It's just the default value refill in initial state. > It's only for ALC668(ALC3661). > The patch was tested pass with issue machine in ODM site. > > Many Thanks. > > BR, > Kailang > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Wednesday, October 22, 2014 4:00 PM > > To: Kailang > > Subject: Re: Restore hidden register for default value > > > > At Wed, 22 Oct 2014 07:56:43 +0000, > > Kailang wrote: > > > > > > Hi Takashi, > > > > > > We want to post ALC668 for hidden register issue. > > > > > > Could I post like below array for codec initial state? > > > > > > WORD coefIndexDefault[][2] = {{0x1, > > 0xbebe},{0x2,0xaaaa},{0x3,0x0},{0x4,0x180},{0x6,0x0}, > > > > > {0x7,0xf80},{0x8,0x31},{0xa,0x60},{0xb,0x0},{0xc,0x7cf7},{0xd, > > 0x1080},{0xe,0x7f7f}, > > > > > {0xf,0xcccc},{0x10,0xddcc},{0x11,0x1},{0x13,0x0},{0x14,0x2aa0} > > ,{0x17,0xA940}, > > > > > {0x19,0},{0x1A,0},{0x1B,0},{0x1c,0x0},{0x1d,0x0},{0x1e,0x7418} > > ,{0x1f,0x804}, > > > > > {0x20,0x4200},{0x21,0x468},{0x22,0x8ccc},{0x23,0x250},{0x24,0x7418}, > > > > > > > > > {0x27,0x0000},{0x28,0x8ccc},{0x2a,0xff00},{0x2b,0x8000},{0xa7,0xff00}} > > > ; Because codec sometimes has wrong value for hidden > register after > > > resume back or reboot. > > > It needed to refill the default value with upper array in > > initial state. > > > But the hidden registers was more for 59 records. > > > Could I post it with no remark? Because it just the default > > value. Thanks. > > > > Some comments would be nice, of course, but not mandatory. > > > > One concern is exactly which codecs should this be applied. > > There are a bunch of codecs compatible with alc668. It's only for > > ALC668, and not for others (ALC668, 670, whatever). > > > > > > Takashi > > > > ------Please consider the environment before printing this e-mail. > >

2 3

[alsa-devel] [PATCH] ASoC:rt286: set combo jack by dmi
by bardliao＠realtek.com 06 Nov '14

06 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch enables combo jack configuration according to dmi. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 97daa80..aa572b6 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1205,6 +1206,16 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); +static struct dmi_system_id intel_platform[] __initdata = { + { + .ident = "Intel Wilson Beach", + .matches = { + DMI_MATCH(DMI_BOARD_NAME, "Wilson Beach SDS") + } + }, + { } +}; + static int rt286_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1240,6 +1251,9 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; + if (dmi_check_system(intel_platform)) + rt286->pdata.cbj_en = true; + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); for (i = 0; i < RT286_POWER_REG_LEN; i++) -- 1.8.1.1.439.g50a6b54

3 2

[alsa-devel] [PATCH] ASoC: samsung: Add MODULE_DEVICE_TABLE for Snow
by Andreas Färber 05 Nov '14

05 Nov '14

This enables the snd_soc_snow module to be auto-loaded. Fixes: da5993df2374 ("ASoC: samsung: Add sound card driver for Snow board") Signed-off-by: Andreas Färber <afaerber(a)suse.de> --- Tested using patches for Spring, but should apply to Snow/Peach-pi equally. Rebased onto sound.git for-next branch. sound/soc/samsung/snow.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 0acf5d0eed53..72118a77dd5b 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -110,6 +110,7 @@ static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98095", }, {}, }; +MODULE_DEVICE_TABLE(of, snow_of_match); static struct platform_driver snow_driver = { .driver = { -- 2.1.2

2 3

[alsa-devel] [PATCH] ASoC: ad193x: Keep DAC output stage active in idle
by Lars-Peter Clausen 05 Nov '14

05 Nov '14

Setting the DAC power-down bit for the ad193x will also disable the DAC output amplifier. This will cause audible clicks and pops when starting or stopping playback. To prevent this a new widget is introduced that controls the DAC power-down bit. This widget is connected to both the DAC and a newly introduced VMID widget. This makes sure that the DAC power-down bit is not set as long as a audio sink is connected to the DAC output. At the same time the PLL and SYSCLK will still be disabled when no playback or capture stream is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 6844d0b..387530b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -72,11 +72,13 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", AD193X_DAC_CTRL0, 0, 1), + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), + SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), @@ -87,13 +89,15 @@ static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, + { "DAC Output", NULL, "DAC" }, + { "DAC Output", NULL, "VMID" }, { "ADC", NULL, "SYSCLK" }, { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", NULL, "DAC" }, - { "DAC2OUT", NULL, "DAC" }, - { "DAC3OUT", NULL, "DAC" }, - { "DAC4OUT", NULL, "DAC" }, + { "DAC1OUT", NULL, "DAC Output" }, + { "DAC2OUT", NULL, "DAC Output" }, + { "DAC3OUT", NULL, "DAC Output" }, + { "DAC4OUT", NULL, "DAC Output" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, { "SYSCLK", NULL, "PLL_PWR" }, -- 1.8.0

2 1

[alsa-devel] [PATCH v2] ALSA: echoaudio: cleanup of unnecessary messages
by Sudip Mukherjee 05 Nov '14

05 Nov '14

commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing chip after it has been freed. This patch fixes that and at the same time removes some debugging messages, which are unnecessary, as they are just printing information about entry and exit from a function, and which switch-case it is executing. we can easily get from ftrace the information about the entry and exit from a function. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change from v1: removing some more debug messages sound/pci/echoaudio/darla20_dsp.c | 2 -- sound/pci/echoaudio/darla24_dsp.c | 2 -- sound/pci/echoaudio/echo3g_dsp.c | 2 -- sound/pci/echoaudio/echoaudio.c | 31 ------------------------------- sound/pci/echoaudio/echoaudio_3g.c | 5 ----- sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------ sound/pci/echoaudio/gina20_dsp.c | 6 ------ sound/pci/echoaudio/gina24_dsp.c | 9 --------- sound/pci/echoaudio/indigo_dsp.c | 2 -- sound/pci/echoaudio/indigodj_dsp.c | 2 -- sound/pci/echoaudio/indigodjx_dsp.c | 2 -- sound/pci/echoaudio/indigoio_dsp.c | 2 -- sound/pci/echoaudio/indigoiox_dsp.c | 2 -- sound/pci/echoaudio/layla20_dsp.c | 9 --------- sound/pci/echoaudio/layla24_dsp.c | 8 -------- sound/pci/echoaudio/mia_dsp.c | 2 -- sound/pci/echoaudio/midi.c | 6 ------ sound/pci/echoaudio/mona_dsp.c | 6 ------ 18 files changed, 110 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index c94e92e..febee5b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; @@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index b1272f88..7b4f6fd 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; @@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index bc37168..ae11ce1 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; @@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1ef29e5..60e4003 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -79,7 +79,6 @@ static void free_firmware(const struct firmware *fw_entry, dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n"); #else release_firmware(fw_entry); - dev_dbg(chip->card->dev, "firmware released\n"); #endif } @@ -96,7 +95,6 @@ static void free_firmware_cache(struct echoaudio *chip) dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -354,7 +352,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -389,7 +386,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -422,7 +418,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -464,7 +459,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -511,7 +505,6 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); @@ -620,7 +613,6 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -691,7 +683,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) } spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -763,10 +754,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -788,9 +777,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -800,7 +787,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -937,7 +923,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital inputs, no outputs */ @@ -950,7 +935,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #else /* ECHOCARD_HAS_VMIXER */ @@ -972,7 +956,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Analog PCM ok\n"); #ifdef ECHOCARD_HAS_DIGITAL_IO /* PCM#1 Digital i/o */ @@ -987,7 +970,6 @@ static int snd_echo_new_pcm(struct echoaudio *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops); if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0) return err; - dev_dbg(chip->card->dev, "Digital PCM ok\n"); #endif /* ECHOCARD_HAS_DIGITAL_IO */ #endif /* ECHOCARD_HAS_VMIXER */ @@ -1881,10 +1863,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1898,14 +1878,12 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1915,7 +1893,6 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -2008,7 +1985,6 @@ static int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return err; } - dev_dbg(card->dev, "Card init OK\n"); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_echo_free(chip); @@ -2038,7 +2014,6 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2191,7 +2166,6 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2218,7 +2192,6 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - dev_dbg(dev, "suspend done\n"); return 0; } @@ -2232,7 +2205,6 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2247,7 +2219,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_free(chip); return err; } - dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2260,7 +2231,6 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2286,7 +2256,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 50a2169..2fa66dc 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -337,13 +336,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Echo3G clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to SPDIF\n"); control_reg |= E3G_SPDIF_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96) control_reg |= E3G_DOUBLE_SPEED_MODE; @@ -353,12 +350,10 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Echo3G clock to ADAT\n"); control_reg |= E3G_ADAT_CLOCK; control_reg &= ~E3G_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Echo3G clock to WORD\n"); control_reg |= E3G_WORD_CLOCK; if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96) control_reg |= E3G_DOUBLE_SPEED_MODE; diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index ba9d4f1..1a9427a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; @@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code) chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); @@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - dev_dbg(chip->card->dev, - "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index a959eae..5dafe92 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; @@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -159,7 +156,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) chip->spdif_status = GD_SPDIF_STATUS_UNDEF; set_sample_rate(chip, chip->sample_rate); chip->input_clock = clock; - dev_dbg(chip->card->dev, "Set Gina clock to INTERNAL\n"); break; case ECHO_CLOCK_SPDIF: chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN; @@ -167,7 +163,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) clear_handshake(chip); send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE); chip->clock_state = GD_CLOCK_SPDIFIN; - dev_dbg(chip->card->dev, "Set Gina20 clock to SPDIF\n"); chip->input_clock = clock; break; default: @@ -209,7 +204,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index c8ea576..6971766 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; @@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & @@ -247,13 +243,11 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Gina24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -263,17 +257,14 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; - dev_dbg(chip->card->dev, "Set Gina24 clock to ADAT\n"); control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC\n"); control_reg |= GML_ESYNC_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_ESYNC96: - dev_dbg(chip->card->dev, "Set Gina24 clock to ESYNC96\n"); control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE; break; default: diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index cdeb073..54edd67 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 133915c..2cf5cc0 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 26cdfcf..5252863 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 5e6df7c..4e81787 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 90cdd27..6de3f9b 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 7f0f6ea..f2024a3 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; @@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -190,24 +188,19 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla20 clock to INTERNAL\n"); rate = chip->sample_rate; clock = LAYLA20_CLOCK_INTERNAL; break; case ECHO_CLOCK_SPDIF: - dev_dbg(chip->card->dev, "Set Layla20 clock to SPDIF\n"); clock = LAYLA20_CLOCK_SPDIF; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Layla20 clock to WORD\n"); clock = LAYLA20_CLOCK_WORD; break; case ECHO_CLOCK_SUPER: - dev_dbg(chip->card->dev, "Set Layla20 clock to SUPER\n"); clock = LAYLA20_CLOCK_SUPER; break; default: @@ -232,7 +225,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -286,7 +278,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index eb8f218..4f11e81 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; @@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -262,7 +258,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Pick the new clock */ switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Layla24 clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -271,7 +266,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_SPDIF_CLOCK; /* Layla24 doesn't support 96KHz S/PDIF */ control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to SPDIF\n"); break; case ECHO_CLOCK_WORD: control_reg |= GML_WORD_CLOCK; @@ -279,14 +273,12 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg |= GML_DOUBLE_SPEED_MODE; else control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to WORD\n"); break; case ECHO_CLOCK_ADAT: if (chip->digital_mode != DIGITAL_MODE_ADAT) return -EAGAIN; control_reg |= GML_ADAT_CLOCK; control_reg &= ~GML_DOUBLE_SPEED_MODE; - dev_dbg(chip->card->dev, "Set Layla24 clock to ADAT\n"); break; default: dev_err(chip->card->dev, diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index ed2f21d..fdad079 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mia\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MIA)) return -ENODEV; @@ -67,7 +66,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 8d43c5a..d913749 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -157,7 +157,6 @@ static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = substream; - dev_dbg(chip->card->dev, "rawmidi_iopen\n"); return 0; } @@ -183,7 +182,6 @@ static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_in = NULL; - dev_dbg(chip->card->dev, "rawmidi_iclose\n"); return 0; } @@ -196,7 +194,6 @@ static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream) chip->tinuse = 0; chip->midi_full = 0; chip->midi_out = substream; - dev_dbg(chip->card->dev, "rawmidi_open\n"); return 0; } @@ -209,7 +206,6 @@ static void snd_echo_midi_output_write(unsigned long data) int bytes, sent, time; unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1]; - dev_dbg(chip->card->dev, "snd_echo_midi_output_write\n"); /* No interrupts are involved: we have to check at regular intervals if the card's output buffer has room for new data. */ sent = bytes = 0; @@ -288,7 +284,6 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream) struct echoaudio *chip = substream->rmidi->private_data; chip->midi_out = NULL; - dev_dbg(chip->card->dev, "rawmidi_oclose\n"); return 0; } @@ -328,6 +323,5 @@ static int snd_echo_midi_create(struct snd_card *card, chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; - dev_dbg(chip->card->dev, "MIDI ok\n"); return 0; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index cc46a8c..843c7a5 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; @@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) @@ -315,7 +312,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) switch (clock) { case ECHO_CLOCK_INTERNAL: - dev_dbg(chip->card->dev, "Set Mona clock to INTERNAL\n"); chip->input_clock = ECHO_CLOCK_INTERNAL; return set_sample_rate(chip, chip->sample_rate); case ECHO_CLOCK_SPDIF: @@ -327,7 +323,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) spin_lock_irq(&chip->lock); if (err < 0) return err; - dev_dbg(chip->card->dev, "Set Mona clock to SPDIF\n"); control_reg |= GML_SPDIF_CLOCK; if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96) control_reg |= GML_DOUBLE_SPEED_MODE; @@ -335,7 +330,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) control_reg &= ~GML_DOUBLE_SPEED_MODE; break; case ECHO_CLOCK_WORD: - dev_dbg(chip->card->dev, "Set Mona clock to WORD\n"); spin_unlock_irq(&chip->lock); err = switch_asic(chip, clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96); -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/6] ASoC: sn95031: Use table based control setup
by Lars-Peter Clausen 05 Nov '14

05 Nov '14

Makes the code a bit cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sn95031.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index cf8fa40..6167c59 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -867,9 +867,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); - snd_soc_add_codec_controls(codec, sn95031_snd_controls, - ARRAY_SIZE(sn95031_snd_controls)); - return 0; } @@ -886,6 +883,9 @@ static struct snd_soc_codec_driver sn95031_codec = { .remove = sn95031_codec_remove, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, + + .controls = sn95031_snd_controls, + .num_controls = ARRAY_SIZE(sn95031_snd_controls), .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), .dapm_routes = sn95031_audio_map, -- 1.8.0

4 10

[alsa-devel] [PATCH] ALSA: usb-audio: Fix device_del() sysfs warnings at disconnect
by Takashi Iwai 05 Nov '14

05 Nov '14

Some USB-audio devices show weird sysfs warnings at disconnecting the devices, e.g. usb 1-3: USB disconnect, device number 3 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 973 at fs/sysfs/group.c:216 device_del+0x39/0x180() sysfs group ffffffff8183df40 not found for kobject 'midiC1D0' Call Trace: [<ffffffff814a3e38>] ? dump_stack+0x49/0x71 [<ffffffff8103cb72>] ? warn_slowpath_common+0x82/0xb0 [<ffffffff8103cc55>] ? warn_slowpath_fmt+0x45/0x50 [<ffffffff813521e9>] ? device_del+0x39/0x180 [<ffffffff81352339>] ? device_unregister+0x9/0x20 [<ffffffff81352384>] ? device_destroy+0x34/0x40 [<ffffffffa00ba29f>] ? snd_unregister_device+0x7f/0xd0 [snd] [<ffffffffa025124e>] ? snd_rawmidi_dev_disconnect+0xce/0x100 [snd_rawmidi] [<ffffffffa00c0192>] ? snd_device_disconnect+0x62/0x90 [snd] [<ffffffffa00c025c>] ? snd_device_disconnect_all+0x3c/0x60 [snd] [<ffffffffa00bb574>] ? snd_card_disconnect+0x124/0x1a0 [snd] [<ffffffffa02e54e8>] ? usb_audio_disconnect+0x88/0x1c0 [snd_usb_audio] [<ffffffffa015260e>] ? usb_unbind_interface+0x5e/0x1b0 [usbcore] [<ffffffff813553e9>] ? __device_release_driver+0x79/0xf0 [<ffffffff81355485>] ? device_release_driver+0x25/0x40 [<ffffffff81354e11>] ? bus_remove_device+0xf1/0x130 [<ffffffff813522b9>] ? device_del+0x109/0x180 [<ffffffffa01501d5>] ? usb_disable_device+0x95/0x1f0 [usbcore] [<ffffffffa014634f>] ? usb_disconnect+0x8f/0x190 [usbcore] [<ffffffffa0149179>] ? hub_thread+0x539/0x13a0 [usbcore] [<ffffffff810669f5>] ? sched_clock_local+0x15/0x80 [<ffffffff81066c98>] ? sched_clock_cpu+0xb8/0xd0 [<ffffffff81070730>] ? bit_waitqueue+0xb0/0xb0 [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffffa0148c40>] ? usb_port_resume+0x430/0x430 [usbcore] [<ffffffff8105973e>] ? kthread+0xce/0xf0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 [<ffffffff814a8b7c>] ? ret_from_fork+0x7c/0xb0 [<ffffffff81059670>] ? kthread_create_on_node+0x1c0/0x1c0 ---[ end trace 40b1928d1136b91e ]--- This comes from the fact that usb-audio driver may receive the disconnect callback multiple times, per each usb interface. When a device has both audio and midi interfaces, it gets called twice, and currently the driver tries to release resources at the last call. At this point, the first parent interface has been already deleted, thus deleting a child of the first parent hits such a warning. For fixing this problem, we need to call snd_card_disconnect() and cancel pending operations at the very first disconnect while the release of the whole objects waits until the last disconnect call. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=80931 Reported-and-tested-by: Tomas Gayoso <tgayoso(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 7ecd0e8a5c51..f61ebb17cc64 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -591,18 +591,19 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, { struct snd_card *card; struct list_head *p; + bool was_shutdown; if (chip == (void *)-1L) return; card = chip->card; down_write(&chip->shutdown_rwsem); + was_shutdown = chip->shutdown; chip->shutdown = 1; up_write(&chip->shutdown_rwsem); mutex_lock(&register_mutex); - chip->num_interfaces--; - if (chip->num_interfaces <= 0) { + if (!was_shutdown) { struct snd_usb_endpoint *ep; snd_card_disconnect(card); @@ -622,6 +623,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, list_for_each(p, &chip->mixer_list) { snd_usb_mixer_disconnect(p); } + } + + chip->num_interfaces--; + if (chip->num_interfaces <= 0) { usb_chip[chip->index] = NULL; mutex_unlock(&register_mutex); snd_card_free_when_closed(card); -- 2.1.3

1 0

[alsa-devel] [RFC] AXD Audio Processing IP ALSA support - Questions
by Qais Yousef 05 Nov '14

05 Nov '14

Hi, I have several questions on the best way to add AXD support in ALSA. The discussion of the previous patch can be found here: https://lkml.org/lkml/2014/10/28/465 Questions: 1- What is the best example to follow to add a simple mp3 support for AXD? The only one I find is in sst-mfld-platform-compress.c in sound/soc/intel directory but it's a bit confusing. I think because it's sharing code with several other sst drivers/platforms. I find the documentation for compress_offload generally lacking. Is there a plan to improve on that? Being a new comer to ALSA framework api, I'm confused what is the correct way to do things :-/ So far I know I need to call snd_soc_register_card(). I thought snd_compress_register() (from compress_driver.h) is how you add compressed nodes to the card but apparently not. It looks like I need to define a compress_dai? Hmmm. 2- Is tinycompress the only userland support for compress_offload? Is there anyone working on gstreamer and omx plugins to support that? Would tinycompress be part of alsa-utils and alsa-libs in the future? I know it needs more work at the moment but it'd be nice if compress_offload support is part of the standard alsa-utils and alsa-libs. 3- Can we get an example of how transcoding (back to disk) is supposed to be working? 4- How can we reconfigure complex audio effect components (like shelving filters) which need filter co-effecient changes to be applied all at once atomically to avoid instability? My plan is to add mp3 driver first then build on that to add the rest of the features. Hopefully I can minimise the amount of work that needs doing (due to me doing it wrongly) and get features upstream sooner. Thanks, Qais

5 9

[alsa-devel] [PATCH] ALSA: echoaudio: cleanup of unnecessary messages
by Sudip Mukherjee 05 Nov '14

05 Nov '14

commit "b5b4a41b392960010fccf1f9ccf8334d612bd450" was dereferencing chip after it has been freed. This patch fixes that and at the same time removes some debugging messages, which are unnecessary, as they are just printing information about entry and exit from a function. This information we can easily get from ftrace. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/echoaudio/darla20_dsp.c | 2 -- sound/pci/echoaudio/darla24_dsp.c | 2 -- sound/pci/echoaudio/echo3g_dsp.c | 2 -- sound/pci/echoaudio/echoaudio.c | 25 ------------------------- sound/pci/echoaudio/echoaudio_3g.c | 1 - sound/pci/echoaudio/echoaudio_dsp.c | 12 ------------ sound/pci/echoaudio/gina20_dsp.c | 4 ---- sound/pci/echoaudio/gina24_dsp.c | 4 ---- sound/pci/echoaudio/indigo_dsp.c | 2 -- sound/pci/echoaudio/indigodj_dsp.c | 2 -- sound/pci/echoaudio/indigodjx_dsp.c | 2 -- sound/pci/echoaudio/indigoio_dsp.c | 2 -- sound/pci/echoaudio/indigoiox_dsp.c | 2 -- sound/pci/echoaudio/layla20_dsp.c | 5 ----- sound/pci/echoaudio/layla24_dsp.c | 4 ---- sound/pci/echoaudio/mona_dsp.c | 3 --- 16 files changed, 74 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index c94e92e..febee5b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA20)) return -ENODEV; @@ -58,7 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index b1272f88..7b4f6fd 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -33,7 +33,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Darla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != DARLA24)) return -ENODEV; @@ -57,7 +56,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw: done\n"); return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index bc37168..ae11ce1 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -46,7 +46,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) int err; local_irq_enable(); - dev_dbg(chip->card->dev, "init_hw() - Echo3G\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != ECHO3G)) return -ENODEV; @@ -99,7 +98,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1ef29e5..146ffc5 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -96,7 +96,6 @@ static void free_firmware_cache(struct echoaudio *chip) dev_dbg(chip->card->dev, "release_firmware(%d)\n", i); } - dev_dbg(chip->card->dev, "firmware_cache released\n"); #endif } @@ -354,7 +353,6 @@ static int pcm_analog_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err; - dev_dbg(chip->card->dev, "pcm_analog_in_open\n"); if ((err = pcm_open(substream, num_analog_busses_in(chip) - substream->number)) < 0) return err; @@ -389,7 +387,6 @@ static int pcm_analog_out_open(struct snd_pcm_substream *substream) #else max_channels = num_analog_busses_out(chip); #endif - dev_dbg(chip->card->dev, "pcm_analog_out_open\n"); if ((err = pcm_open(substream, max_channels - substream->number)) < 0) return err; if ((err = snd_pcm_hw_rule_add(substream->runtime, 0, @@ -422,7 +419,6 @@ static int pcm_digital_in_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_in_open\n"); max_channels = num_digital_busses_in(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -464,7 +460,6 @@ static int pcm_digital_out_open(struct snd_pcm_substream *substream) struct echoaudio *chip = snd_pcm_substream_chip(substream); int err, max_channels; - dev_dbg(chip->card->dev, "pcm_digital_out_open\n"); max_channels = num_digital_busses_out(chip) - substream->number; mutex_lock(&chip->mode_mutex); if (chip->digital_mode == DIGITAL_MODE_ADAT) @@ -511,7 +506,6 @@ static int pcm_close(struct snd_pcm_substream *substream) /* Nothing to do here. Audio is already off and pipe will be * freed by its callback */ - dev_dbg(chip->card->dev, "pcm_close\n"); atomic_dec(&chip->opencount); oc = atomic_read(&chip->opencount); @@ -620,7 +614,6 @@ static int init_engine(struct snd_pcm_substream *substream, spin_lock_irq(&chip->lock); set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den); spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_params ok\n"); return 0; } @@ -691,7 +684,6 @@ static int pcm_hw_free(struct snd_pcm_substream *substream) } spin_unlock_irq(&chip->lock); - dev_dbg(chip->card->dev, "pcm_hw_freed\n"); snd_pcm_lib_free_pages(substream); return 0; } @@ -763,10 +755,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - dev_dbg(chip->card->dev, "pcm_trigger resume\n"); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dev_dbg(chip->card->dev, "pcm_trigger start\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -788,9 +778,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) chip->pipe_cyclic_mask); break; case SNDRV_PCM_TRIGGER_SUSPEND: - dev_dbg(chip->card->dev, "pcm_trigger suspend\n"); case SNDRV_PCM_TRIGGER_STOP: - dev_dbg(chip->card->dev, "pcm_trigger stop\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -800,7 +788,6 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = stop_transport(chip, channelmask); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev_dbg(chip->card->dev, "pcm_trigger pause\n"); for (i = 0; i < DSP_MAXPIPES; i++) { if (channelmask & (1 << i)) { pipe = chip->substream[i]->runtime->private_data; @@ -1881,10 +1868,8 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) static int snd_echo_free(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "Stop DSP...\n"); if (chip->comm_page) rest_in_peace(chip); - dev_dbg(chip->card->dev, "Stopped.\n"); if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1898,14 +1883,12 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->iores) release_and_free_resource(chip->iores); - dev_dbg(chip->card->dev, "MMIO freed.\n"); pci_disable_device(chip->pci); /* release chip data */ free_firmware_cache(chip); kfree(chip); - dev_dbg(chip->card->dev, "Chip freed.\n"); return 0; } @@ -1915,7 +1898,6 @@ static int snd_echo_dev_free(struct snd_device *device) { struct echoaudio *chip = device->device_data; - dev_dbg(chip->card->dev, "snd_echo_dev_free()...\n"); return snd_echo_free(chip); } @@ -2038,7 +2020,6 @@ static int snd_echo_probe(struct pci_dev *pci, return -ENOENT; } - dev_dbg(&pci->dev, "Echoaudio driver starting...\n"); i = 0; err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); @@ -2191,7 +2172,6 @@ static int snd_echo_suspend(struct device *dev) struct pci_dev *pci = to_pci_dev(dev); struct echoaudio *chip = dev_get_drvdata(dev); - dev_dbg(dev, "suspend start\n"); snd_pcm_suspend_all(chip->analog_pcm); snd_pcm_suspend_all(chip->digital_pcm); @@ -2218,7 +2198,6 @@ static int snd_echo_suspend(struct device *dev) pci_save_state(pci); pci_disable_device(pci); - dev_dbg(dev, "suspend done\n"); return 0; } @@ -2232,7 +2211,6 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - dev_dbg(dev, "resume start\n"); pci_restore_state(pci); commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); if (commpage_bak == NULL) @@ -2247,7 +2225,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_free(chip); return err; } - dev_dbg(dev, "resume init OK\n"); /* Temporarily set chip->pipe_alloc_mask=0 otherwise * restore_dsp_settings() fails. @@ -2260,7 +2237,6 @@ static int snd_echo_resume(struct device *dev) kfree(commpage_bak); return err; } - dev_dbg(dev, "resume restore OK\n"); memcpy(&commpage->audio_format, &commpage_bak->audio_format, sizeof(commpage->audio_format)); @@ -2286,7 +2262,6 @@ static int snd_echo_resume(struct device *dev) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif - dev_dbg(dev, "resume done\n"); return 0; } diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 50a2169..7258328 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -328,7 +328,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index ba9d4f1..1a9427a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,7 +206,6 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) goto la_error; } - dev_dbg(chip->card->dev, "ASIC loaded\n"); free_firmware(fw, chip); return 0; @@ -473,7 +472,6 @@ static int load_dsp(struct echoaudio *chip, u16 *code) chip->dsp_code = code; /* Show which DSP code loaded */ chip->bad_board = FALSE; /* DSP OK */ - dev_dbg(chip->card->dev, "load_dsp: OK!\n"); return 0; } udelay(100); @@ -674,7 +672,6 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { int i, o, err; - dev_dbg(chip->card->dev, "restore_dsp_settings\n"); if ((err = check_asic_status(chip)) < 0) return err; @@ -771,7 +768,6 @@ static int restore_dsp_rettings(struct echoaudio *chip) if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) return -EIO; - dev_dbg(chip->card->dev, "restore_dsp_rettings done\n"); return 0; } @@ -865,7 +861,6 @@ Same thing for pause_ and stop_ -trasport below. */ static int start_transport(struct echoaudio *chip, u32 channel_mask, u32 cyclic_mask) { - dev_dbg(chip->card->dev, "start_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -891,7 +886,6 @@ static int start_transport(struct echoaudio *chip, u32 channel_mask, static int pause_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "pause_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -918,7 +912,6 @@ static int pause_transport(struct echoaudio *chip, u32 channel_mask) static int stop_transport(struct echoaudio *chip, u32 channel_mask) { - dev_dbg(chip->card->dev, "stop_transport %x\n", channel_mask); if (wait_handshake(chip)) return -EIO; @@ -954,8 +947,6 @@ static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index) stopped and unallocated. */ static int rest_in_peace(struct echoaudio *chip) { - dev_dbg(chip->card->dev, - "rest_in_peace() open=%x\n", chip->pipe_alloc_mask); /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); @@ -1018,7 +1009,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) */ static int init_line_levels(struct echoaudio *chip) { - dev_dbg(chip->card->dev, "init_line_levels\n"); memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); @@ -1099,7 +1089,6 @@ static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe, it moves data. The DMA counter is in units of bytes, not samples. */ pipe->dma_counter = &chip->comm_page->position[pipe_index]; *pipe->dma_counter = 0; - dev_dbg(chip->card->dev, "allocate_pipes: ok\n"); return pipe_index; } @@ -1110,7 +1099,6 @@ static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe) u32 channel_mask; int i; - dev_dbg(chip->card->dev, "free_pipes: Pipe %d\n", pipe->index); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe->index))) return -EINVAL; if (snd_BUG_ON(pipe->state != PIPE_STATE_STOPPED)) diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index a959eae..a8dcf1e 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -37,7 +37,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA20)) return -ENODEV; @@ -63,7 +62,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -150,7 +148,6 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) static int set_input_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_input_clock:\n"); switch (clock) { case ECHO_CLOCK_INTERNAL: @@ -209,7 +206,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index c8ea576..165fe3d 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Gina24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != GINA24)) return -ENODEV; @@ -79,7 +78,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -156,7 +154,6 @@ static int load_asic(struct echoaudio *chip) control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; err = write_control_reg(chip, control_reg, TRUE); } - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } @@ -238,7 +235,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) { u32 control_reg, clocks_from_dsp; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Mask off the clock select bits */ control_reg = le32_to_cpu(chip->comm_page->control_register) & diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index cdeb073..54edd67 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 133915c..2cf5cc0 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJ\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJ)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 26cdfcf..5252863 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo DJx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 5e6df7c..4e81787 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -38,7 +38,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IO\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IO)) return -ENODEV; @@ -61,7 +60,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 90cdd27..6de3f9b 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -35,7 +35,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Indigo IOx\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) return -ENODEV; @@ -60,7 +59,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 7f0f6ea..637e72f 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla20\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA20)) return -ENODEV; @@ -65,7 +64,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -190,7 +188,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) u16 clock; u32 rate; - dev_dbg(chip->card->dev, "set_input_clock:\n"); rate = 0; switch (clock_source) { case ECHO_CLOCK_INTERNAL: @@ -232,7 +229,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock_source) static int set_output_clock(struct echoaudio *chip, u16 clock) { - dev_dbg(chip->card->dev, "set_output_clock: %d\n", clock); switch (clock) { case ECHO_CLOCK_SUPER: clock = LAYLA20_OUTPUT_CLOCK_SUPER; @@ -286,7 +282,6 @@ static int update_flags(struct echoaudio *chip) static int set_professional_spdif(struct echoaudio *chip, char prof) { - dev_dbg(chip->card->dev, "set_professional_spdif %d\n", prof); if (prof) chip->comm_page->flags |= cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index eb8f218..93813a8 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -40,7 +40,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Layla24\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != LAYLA24)) return -ENODEV; @@ -70,7 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -118,7 +116,6 @@ static int load_asic(struct echoaudio *chip) if (chip->asic_loaded) return 1; - dev_dbg(chip->card->dev, "load_asic\n"); /* Give the DSP a few milliseconds to settle down */ mdelay(10); @@ -152,7 +149,6 @@ static int load_asic(struct echoaudio *chip) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, TRUE); - dev_dbg(chip->card->dev, "load_asic() done\n"); return err; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index cc46a8c..63a6a50 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -41,7 +41,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) { int err; - dev_dbg(chip->card->dev, "init_hw() - Mona\n"); if (snd_BUG_ON((subdevice_id & 0xfff0) != MONA)) return -ENODEV; @@ -72,7 +71,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - dev_dbg(chip->card->dev, "init_hw done\n"); return err; } @@ -302,7 +300,6 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) u32 control_reg, clocks_from_dsp; int err; - dev_dbg(chip->card->dev, "set_input_clock:\n"); /* Prevent two simultaneous calls to switch_asic() */ if (atomic_read(&chip->opencount)) -- 1.8.1.2

3 4

[alsa-devel] [PATCH v2 0/5] ASoC: Intel: ACPI support and machines
by Vinod Koul 05 Nov '14

05 Nov '14

This is second rev of patch series which fixes the modules split as discussed in ML. The core part is now module along with PCI and ACPI one. Added fix reported by Dan as well. Also added the BYTCR machine driver Subhransu S. Prusty (1): ASoC: Intel: add BYTCR machine driver with RT5640 Vinod Koul (4): ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld- add ACPI module sound/soc/intel/Kconfig | 22 ++- sound/soc/intel/Makefile | 2 + sound/soc/intel/bytcr_dpcm_rt5640.c | 258 +++++++++++++++++++++++ sound/soc/intel/sst-atom-controls.c | 2 +- sound/soc/intel/sst/Makefile | 8 +- sound/soc/intel/sst/sst.c | 243 +++++----------------- sound/soc/intel/sst/sst.h | 9 +- sound/soc/intel/sst/sst_acpi.c | 386 +++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pci.c | 225 ++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 10 files changed, 958 insertions(+), 199 deletions(-) create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_pci.c

4 18

[alsa-devel] [PATCH] ASoC: Intel: Fix the driver data not set issue
by Jie Yang 05 Nov '14

05 Nov '14

The priv_data is allocated again here wrongly, and it is not set to the driver data after assignment. This make the pdata->dev is NULL and oops occurs on the first call to hsw_volume_put. The resource has been allocated in driver probe callback hsw_pcm_dev_probe, so here just remove this sencond allocation is OK. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 21f1316..bc02a6e 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -866,7 +866,6 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) dev = platform->dev; dma_dev = pdata->dma_dev; - priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); priv_data->hsw = pdata->dsp; priv_data->dev = platform->dev; priv_data->pm_state = HSW_PM_STATE_D0; -- 1.9.1

3 3

[alsa-devel] [PATCH] ALSA: hda - fix mute led problem for three HP laptops
by Hui Wang 05 Nov '14

05 Nov '14

Without the fix, the mute led can't work on these three machines. After apply this fix, these three machines will fall back on the led control quirk as below, and through testing, the mute led works very well. PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC282_STANDARD_PINS, {0x12, 0x90a60140}, ... BugLink: https://bugs.launchpad.net/bugs/1389497 Tested-by: TieFu Chen <tienfu.chen(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ba72fff..1af917f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4744,9 +4744,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2247, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2248, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2249, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), -- 1.9.1

2 1

[alsa-devel] Compressed audio support over HDMI
by D K 05 Nov '14

05 Nov '14

Hi, I recently came across the compressed audio framework used with ALSA-SOC. Can the framework be used to send compressed audio to HDMI driver?.Like using tinycompress from usespace . thanks,

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Use specific r/w function for DSP mode
by Oder Chiou 04 Nov '14

04 Nov '14

In DSP mode, the register r/w should use the specific function to access that is invoked by address mapping of the DSP. The MX-65[1] is for switching DSP or codec mode. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 222 ++++++++++++++++++++++++++++++---------------- sound/soc/codecs/rt5677.h | 2 + 2 files changed, 149 insertions(+), 75 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 413bccb..74f65ff 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -539,53 +539,113 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } } +static int rt5677_hw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + u8 r[1]; + u8 data[2]; + int ret; + + /* Write register */ + r[0] = reg & 0xff; + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = &r[0]; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return -EIO; + } + + *val = (data[0] << 8) | data[1]; + + dev_dbg(&client->dev, "read %02x => %04x\n", reg, *val); + + return 0; +} + +static int rt5677_hw_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + u8 data[3]; + int ret; + + data[0] = reg & 0xff; + data[1] = (0xff00 & val) >> 8; + data[2] = val & 0xff; + + dev_dbg(&client->dev, "write %02x = %04x\n", reg, val); + + ret = i2c_master_send(client, data, 3); + if (ret == 3) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + /** * rt5677_dsp_mode_i2c_write_addr - Write value to address on DSP mode. - * @codec: SoC audio codec device. + * @rt5677: Private Data. * @addr: Address index. * @value: Address data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write_addr(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write_addr(struct rt5677_priv *rt5677, unsigned int addr, unsigned int value, unsigned int opcode) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt5677->i2c; int ret; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_DATA_MSB, value >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set data msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set data msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_DATA_LSB, value & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set data lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set data lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE, opcode); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_OP_CODE, opcode); if (ret < 0) { - dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + dev_err(&client->dev, "Failed to set op code value: %d\n", + ret); goto err; } @@ -597,42 +657,45 @@ err: /** * rt5677_dsp_mode_i2c_read_addr - Read value from address on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @addr: Address index. * @value: Address data. * + * * Returns 0 for success or negative error code. */ static int rt5677_dsp_mode_i2c_read_addr( - struct snd_soc_codec *codec, unsigned int addr, unsigned int *value) + struct rt5677_priv *rt5677, unsigned int addr, unsigned int *value) { - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *client = rt5677->i2c; int ret; unsigned int msb, lsb; mutex_lock(&rt5677->dsp_cmd_lock); - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_MSB, addr >> 16); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr msb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr msb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_ADDR_LSB, + ret = rt5677_hw_write(client, RT5677_DSP_I2C_ADDR_LSB, addr & 0xffff); if (ret < 0) { - dev_err(codec->dev, "Failed to set addr lsb value: %d\n", ret); + dev_err(&client->dev, "Failed to set addr lsb value: %d\n", + ret); goto err; } - ret = regmap_write(rt5677->regmap, RT5677_DSP_I2C_OP_CODE , 0x0002); + ret = rt5677_hw_write(client, RT5677_DSP_I2C_OP_CODE , 0x0002); if (ret < 0) { - dev_err(codec->dev, "Failed to set op code value: %d\n", ret); + dev_err(&client->dev, "Failed to set op code value: %d\n", ret); goto err; } - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_MSB, &msb); - regmap_read(rt5677->regmap, RT5677_DSP_I2C_DATA_LSB, &lsb); + rt5677_hw_read(client, RT5677_DSP_I2C_DATA_MSB, &msb); + rt5677_hw_read(client, RT5677_DSP_I2C_DATA_LSB, &lsb); *value = (msb << 16) | lsb; err: @@ -643,17 +706,17 @@ err: /** * rt5677_dsp_mode_i2c_write - Write register on DSP mode. - * @codec: SoC audio codec device. + * rt5677: Private Data. * @reg: Register index. * @value: Register data. * * * Returns 0 for success or negative error code. */ -static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, +static int rt5677_dsp_mode_i2c_write(struct rt5677_priv *rt5677, unsigned int reg, unsigned int value) { - return rt5677_dsp_mode_i2c_write_addr(codec, 0x18020000 + reg * 2, + return rt5677_dsp_mode_i2c_write_addr(rt5677, 0x18020000 + reg * 2, value, 0x0001); } @@ -661,57 +724,33 @@ static int rt5677_dsp_mode_i2c_write(struct snd_soc_codec *codec, * rt5677_dsp_mode_i2c_read - Read register on DSP mode. * @codec: SoC audio codec device. * @reg: Register index. + * @value: Register data. * * - * Returns Register value. + * Returns 0 for success or negative error code. */ -static unsigned int rt5677_dsp_mode_i2c_read( - struct snd_soc_codec *codec, unsigned int reg) +static int rt5677_dsp_mode_i2c_read( + struct rt5677_priv *rt5677, unsigned int reg, unsigned int *value) { - unsigned int value = 0; + int ret = rt5677_dsp_mode_i2c_read_addr(rt5677, 0x18020000 + reg * 2, + value); - rt5677_dsp_mode_i2c_read_addr(codec, 0x18020000 + reg * 2, &value); + *value &= 0xffff; - return value; + return ret; } -/** - * rt5677_dsp_mode_i2c_update_bits - update register on DSP mode. - * @codec: audio codec - * @reg: register index. - * @mask: register mask - * @value: new value - * - * - * Returns 1 for change, 0 for no change, or negative error code. - */ -static int rt5677_dsp_mode_i2c_update_bits(struct snd_soc_codec *codec, - unsigned int reg, unsigned int mask, unsigned int value) +static void rt5677_set_dsp_mode(struct snd_soc_codec *codec, bool on) { - unsigned int old, new; - int change, ret; - - ret = rt5677_dsp_mode_i2c_read(codec, reg); - if (ret < 0) { - dev_err(codec->dev, "Failed to read reg: %d\n", ret); - goto err; - } + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = rt5677_dsp_mode_i2c_write(codec, reg, new); - if (ret < 0) { - dev_err(codec->dev, - "Failed to write reg: %d\n", ret); - goto err; - } + if (on) { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x2); + rt5677->is_dsp_mode = true; + } else { + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x2, 0x0); + rt5677->is_dsp_mode = false; } - return change; - -err: - return ret; } static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) @@ -733,9 +772,14 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_write(rt5677->regmap, RT5677_GLB_CLK2, 0x0080); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); - regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07ff); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); + rt5677_set_dsp_mode(codec, true); ret = request_firmware(&rt5677->fw1, RT5677_FIRMWARE1, codec->dev); @@ -751,8 +795,7 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) release_firmware(rt5677->fw2); } - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x0); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x0); regcache_cache_bypass(rt5677->regmap, false); regcache_cache_only(rt5677->regmap, true); @@ -762,9 +805,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) regcache_cache_only(rt5677->regmap, false); regcache_cache_bypass(rt5677->regmap, true); - rt5677_dsp_mode_i2c_update_bits(codec, RT5677_PWR_DSP1, 0x1, - 0x1); - rt5677_dsp_mode_i2c_write(codec, RT5677_PWR_DSP1, 0x0001); + regmap_update_bits(rt5677->regmap, RT5677_PWR_DSP1, 0x1, 0x1); + rt5677_set_dsp_mode(codec, false); + regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x0001); regmap_write(rt5677->regmap, RT5677_RESET, 0x10ec); @@ -3804,6 +3847,32 @@ static int rt5677_resume(struct snd_soc_codec *codec) #define rt5677_resume NULL #endif +static int rt5677_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + else + rt5677_hw_read(context, reg, val); + + return 0; +} + +static int rt5677_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5677_priv *rt5677 = i2c_get_clientdata(client); + + if (rt5677->is_dsp_mode) + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + else + rt5677_hw_write(context, reg, val); + + return 0; +} + #define RT5677_STEREO_RATES SNDRV_PCM_RATE_8000_96000 #define RT5677_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) @@ -3938,6 +4007,8 @@ static const struct regmap_config rt5677_regmap = { .volatile_reg = rt5677_volatile_register, .readable_reg = rt5677_readable_register, + .reg_read = rt5677_read, + .reg_write = rt5677_write, .cache_type = REGCACHE_RBTREE, .reg_defaults = rt5677_reg, @@ -4063,6 +4134,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, rt5677); + rt5677->i2c = i2c; if (pdata) rt5677->pdata = *pdata; @@ -4094,7 +4166,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, msleep(10); } - rt5677->regmap = devm_regmap_init_i2c(i2c, &rt5677_regmap); + rt5677->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5677_regmap); if (IS_ERR(rt5677->regmap)) { ret = PTR_ERR(rt5677->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index d2c743c..ab81659f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1599,6 +1599,7 @@ struct rt5677_priv { struct regmap *regmap; const struct firmware *fw1, *fw2; struct mutex dsp_cmd_lock; + struct i2c_client *i2c; int sysclk; int sysclk_src; @@ -1614,6 +1615,7 @@ struct rt5677_priv { #endif bool dsp_vad_en; struct regmap_irq_chip_data *irq_data; + bool is_dsp_mode; }; #endif /* __RT5677_H__ */ -- 1.8.1.1.439.g50a6b54

2 5

[alsa-devel] [PATCH 1/2] ASoC: max98090: Correct pclk divisor settings
by Dylan Reid 04 Nov '14

04 Nov '14

The Baytrail-based chromebooks have a 20MHz mclk, the code was setting the divisor incorrectly in this case. According to the 98090 datasheet, the divisor should be set to DIV1 for 10 <= mclk <= 20. Correct this and the surrounding clock ranges as well to match the datasheet. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/max98090.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index d519294..1229554 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1941,13 +1941,13 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai, * 0x02 (when master clk is 20MHz to 40MHz).. * 0x03 (when master clk is 40MHz to 60MHz).. */ - if ((freq >= 10000000) && (freq < 20000000)) { + if ((freq >= 10000000) && (freq <= 20000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV1); - } else if ((freq >= 20000000) && (freq < 40000000)) { + } else if ((freq > 20000000) && (freq <= 40000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV2); - } else if ((freq >= 40000000) && (freq < 60000000)) { + } else if ((freq > 40000000) && (freq <= 60000000)) { snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK, M98090_PSCLK_DIV4); } else { -- 2.1.2.330.g565301e

2 2

[alsa-devel] [PATCH v6] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 04 Nov '14

04 Nov '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 143 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 146 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471..3ee031e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..51c6c4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -906,6 +906,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d8..1db2168 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1461,7 +1461,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1473,7 +1474,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cdc837e..68f6057 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -729,6 +729,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2664,6 +2694,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3142,6 +3175,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config += w->params_select; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3222,8 +3258,35 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3231,14 +3294,48 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = kcalloc(num_params, sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + w_param_text[count] = kmemdup((void *)(config->stream_name), + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3246,6 +3343,25 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + private_value = + (unsigned long) kmemdup((void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + template.kcontrol_news = kmemdup(&kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3253,15 +3369,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dpcm: Fix race between FE/BE updates and trigger
by Takashi Iwai 04 Nov '14

04 Nov '14

DPCM can update the FE/BE connection states totally asynchronously from the FE's PCM state. Most of FE/BE state changes are protected by mutex, so that they won't race, but there are still some actions that are uncovered. For example, suppose to switch a BE while a FE's stream is running. This would call soc_dpcm_runtime_update(), which sets FE's runtime_update flag, then sets up and starts BEs, and clears FE's runtime_update flag again. When a device emits XRUN during this operation, the PCM core triggers snd_pcm_stop(XRUN). Since the trigger action is an atomic ops, this isn't blocked by the mutex, thus it kicks off DPCM's trigger action. It eventually updates and clears FE's runtime_update flag while soc_dpcm_runtime_update() is running concurrently, and it results in confusion. Usually, for avoiding such a race, we take a lock. There is a PCM stream lock for that purpose. However, as already mentioned, the trigger action is atomic, and we can't take the lock for the whole soc_dpcm_runtime_update() or other operations that include the lengthy jobs like hw_params or prepare. This patch provides an alternative solution. This adds a way to defer the conflicting trigger callback to be executed at the end of FE/BE state changes. For doing it, two things are introduced: - Each runtime_update state change of FEs is protected via PCM stream lock. - The FE's trigger callback checks the runtime_update flag. If it's not set, the trigger action is executed there. If set, mark the pending trigger action and returns immediately. - At the exit of runtime_update state change, it checks whether the pending trigger is present. If yes, it executes the trigger action at this point. Reported-and-tested-by: Qiao Zhou <zhouqiao(a)marvell.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc-dpcm.h | 2 ++ sound/soc/soc-pcm.c | 72 +++++++++++++++++++++++++++++++++++++----------- 2 files changed, 58 insertions(+), 16 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 2883a7a6f9f3..98f2ade0266e 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -102,6 +102,8 @@ struct snd_soc_dpcm_runtime { /* state and update */ enum snd_soc_dpcm_update runtime_update; enum snd_soc_dpcm_state state; + + int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; /* can this BE stop and free */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311afdeaa..57277dd79e11 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1522,13 +1522,36 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); } +static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd); + +/* Set FE's runtime_update state; the state is protected via PCM stream lock + * for avoiding the race with trigger callback. + * If the state is unset and a trigger is pending while the previous operation, + * process the pending trigger action here. + */ +static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, + int stream, enum snd_soc_dpcm_update state) +{ + struct snd_pcm_substream *substream = + snd_soc_dpcm_get_substream(fe, stream); + + snd_pcm_stream_lock_irq(substream); + if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { + dpcm_fe_dai_do_trigger(substream, + fe->dpcm[stream].trigger_pending - 1); + fe->dpcm[stream].trigger_pending = 0; + } + fe->dpcm[stream].runtime_update = state; + snd_pcm_stream_unlock_irq(substream); +} + static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_pcm_runtime *runtime = fe_substream->runtime; int stream = fe_substream->stream, ret = 0; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); ret = dpcm_be_dai_startup(fe, fe_substream->stream); if (ret < 0) { @@ -1550,13 +1573,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_runtime(fe_substream); snd_pcm_limit_hw_rates(runtime); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; unwind: dpcm_be_dai_startup_unwind(fe, fe_substream->stream); be_err: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } @@ -1603,7 +1626,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ dpcm_be_dai_shutdown(fe, substream->stream); @@ -1617,7 +1640,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; } @@ -1665,7 +1688,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) int err, stream = substream->stream; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); dev_dbg(fe->dev, "ASoC: hw_free FE %s\n", fe->dai_link->name); @@ -1680,7 +1703,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) err = dpcm_be_dai_hw_free(fe, stream); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return 0; @@ -1773,7 +1796,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, int ret, stream = substream->stream; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); memcpy(&fe->dpcm[substream->stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); @@ -1796,7 +1819,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS; out: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return ret; } @@ -1910,7 +1933,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream, ret; @@ -1984,6 +2007,23 @@ out: return ret; } +static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + int stream = substream->stream; + + /* if FE's runtime_update is already set, we're in race; + * process this trigger later at exit + */ + if (fe->dpcm[stream].runtime_update != SND_SOC_DPCM_UPDATE_NO) { + fe->dpcm[stream].trigger_pending = cmd + 1; + return 0; /* delayed, assuming it's successful */ + } + + /* we're alone, let's trigger */ + return dpcm_fe_dai_do_trigger(substream, cmd); +} + int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -2027,7 +2067,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) dev_dbg(fe->dev, "ASoC: prepare FE %s\n", fe->dai_link->name); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { @@ -2054,7 +2094,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); mutex_unlock(&fe->card->mutex); return ret; @@ -2201,11 +2241,11 @@ static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream) { int ret; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); ret = dpcm_run_update_startup(fe, stream); if (ret < 0) dev_err(fe->dev, "ASoC: failed to startup some BEs\n"); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } @@ -2214,11 +2254,11 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) { int ret; - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE); ret = dpcm_run_update_shutdown(fe, stream); if (ret < 0) dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n"); - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; } -- 2.1.3

3 2

Re: [alsa-devel] ALSA: echoaudio: remove all snd_printk
by Dan Carpenter 04 Nov '14

04 Nov '14

Hello Sudip Mukherjee, The patch b5b4a41b3929: "ALSA: echoaudio: remove all snd_printk" from Nov 3, 2014, leads to the following static checker warning: sound/pci/echoaudio/echoaudio.c:1908 snd_echo_free() error: dereferencing freed memory 'chip' sound/pci/echoaudio/echoaudio.c 1905 /* release chip data */ 1906 free_firmware_cache(chip); 1907 kfree(chip); ^^^^^^^^^^^ 1908 dev_dbg(chip->card->dev, "Chip freed.\n"); ^^^^^^^^^^ We have lots of printks in this function. We could just delete them all and no one would care. These days you can get the same information from ftrace. 1909 return 0; 1910 } regards, dan carpenter

3 2

[alsa-devel] [PATCH] ASoC: Intel: Fix build with CONFIG_SLEEP enabled.
by Liam Girdwood 04 Nov '14

04 Nov '14

Fix the following build error when CONFIG_SLEEP is enabled and CONFIG_RUNTIME is disabled. The BDW ADSP sleep PM functionality depends on the runtime pm calls for context save/restore. All error/warnings: >> ERROR: "snd_soc_suspend" undefined! >> ERROR: "snd_soc_resume" undefined! Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 3cb9ed7..afbcc9b 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -1011,7 +1011,6 @@ static int hsw_pcm_dev_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM #ifdef CONFIG_PM_RUNTIME static int hsw_pcm_runtime_idle(struct device *dev) @@ -1071,6 +1070,8 @@ static int hsw_pcm_runtime_resume(struct device *dev) #define hsw_pcm_runtime_resume NULL #endif +#if defined(CONFIG_PM_SLEEP) && defined(CONFIG_PM_RUNTIME) + static void hsw_pcm_complete(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1166,6 +1167,11 @@ static int hsw_pcm_prepare(struct device *dev) return 0; } +#else +#define hsw_pcm_prepare NULL +#define hsw_pcm_complete NULL +#endif + static const struct dev_pm_ops hsw_pcm_pm = { .runtime_idle = hsw_pcm_runtime_idle, .runtime_suspend = hsw_pcm_runtime_suspend, @@ -1173,9 +1179,6 @@ static const struct dev_pm_ops hsw_pcm_pm = { .prepare = hsw_pcm_prepare, .complete = hsw_pcm_complete, }; -#else -#define hsw_pcm_pm NULL -#endif static struct platform_driver hsw_pcm_driver = { .driver = { -- 2.1.0

2 1

[alsa-devel] [PATCH 0/5] A few ALSA PCM core update patches
by Takashi Iwai 04 Nov '14

04 Nov '14

Hi, here is a small series of patches for cleaning up / enhancing the PCM core stuff. [PATCH 1/5] ALSA: pcm: Simplify snd_pcm_action_lock_irq() [PATCH 2/5] ALSA: pcm: Refactoring snd_pcm_action() [PATCH 3/5] ALSA: pcm: Correct PCM BUG error message [PATCH 4/5] ALSA: pcm: Replace PCM hwptr tracking with tracepoints [PATCH 5/5] ALSA: pcm: Add xrun_injection proc entry Takashi

1 5

[alsa-devel] ASoC: no dapm match for AMIC
by Fabio Estevam 04 Nov '14

04 Nov '14

Hi, I am running linux-next 20141104 on a imx6q-sabresd board and noticed the following error messages: [ 2.145602] imx-wm8962 sound: wm8962 <-> 202c000.ssi mapping ok [ 2.151654] imx-wm8962 sound: Connecting non-supply widget to supply widget is not supported (AMIC -> MICBIAS) [ 2.161710] imx-wm8962 sound: ASoC: no dapm match for AMIC --> (null) --> MICBIAS [ 2.169258] imx-wm8962 sound: ASoC: Failed to add route AMIC -> direct -> MICBIAS What would be the correct way to fix them? Thanks

2 2

[alsa-devel] [PATCH 1/2] ASoC: Remove card field from snd_soc_dai struct
by Lars-Peter Clausen 04 Nov '14

04 Nov '14

The card field of the snd_soc_dai field is very rarely used. We can use dai->component->card instead and remove the card field from the snd_soc_dai struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dai.h | 2 -- sound/soc/soc-core.c | 7 +------ sound/soc/soc-dapm.c | 2 +- 3 files changed, 2 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 45d0fa1..373d177 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -275,8 +275,6 @@ struct snd_soc_dai { unsigned int tx_mask; unsigned int rx_mask; - struct snd_soc_card *card; - struct list_head list; }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3818cf3..d9b7b70 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1315,11 +1315,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, num, order); - /* config components */ - cpu_dai->card = card; - for (i = 0; i < rtd->num_codecs; i++) - rtd->codec_dais[i]->card = card; - /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -2314,7 +2309,7 @@ EXPORT_SYMBOL_GPL(snd_soc_add_card_controls); int snd_soc_add_dai_controls(struct snd_soc_dai *dai, const struct snd_kcontrol_new *controls, int num_controls) { - struct snd_card *card = dai->card->snd_card; + struct snd_card *card = dai->component->card->snd_card; return snd_soc_add_controls(card, dai->dev, controls, num_controls, NULL, dai); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6bf2c97..c5136bb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1043,7 +1043,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget_list **list) { - struct snd_soc_card *card = dai->card; + struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; int paths; -- 1.8.0

2 3

[alsa-devel] [PATCH v5 0/3] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 04 Nov '14

04 Nov '14

By mistake previous patchset versions was sent as in-reply-to to previous patchset versions, so they may have been lost among other mails. One of pathces was already applied by Mark Brown, here is 3 patches implementing audio support on Phytec phyFLEX-i.MX6 boards. This patchset version only renames audio card name to OnboardTLV320AIC3007 (on request from Phytec). Patches are based on branch for-next of git://git.kernel.org/pub/scm/linux/kernel/git/shawnguo/linux.git

3 6

[alsa-devel] [PATCH] ASoC: rt5645: make bias level more reasonale
by bardliao＠realtek.com 04 Nov '14

04 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc442f4..0e92e63 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2071,8 +2071,8 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { - case SND_SOC_BIAS_STANDBY: - if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2, @@ -2087,15 +2087,24 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, } break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2, + RT5645_PWR_FV1 | RT5645_PWR_FV2); + break; + case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); snd_soc_write(codec, RT5645_GEN_CTRL1, 0x0128); - snd_soc_write(codec, RT5645_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5645_PWR_DIG2, 0x0000); - snd_soc_write(codec, RT5645_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5645_PWR_MIXER, 0x0000); - snd_soc_write(codec, RT5645_PWR_ANLG1, 0x0000); - snd_soc_write(codec, RT5645_PWR_ANLG2, 0x0000); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2 | + RT5645_PWR_FV1 | RT5645_PWR_FV2, 0x0); break; default: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt286: reduce power consumption
by bardliao＠realtek.com 04 Nov '14

04 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch will optimize the power consumption of rt286. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 211 ++++++++++++++++++++++++++++++++++------------- 1 file changed, 155 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 4aa555c..97daa80 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -36,11 +36,13 @@ struct rt286_priv { struct regmap *regmap; + struct snd_soc_codec *codec; struct rt286_platform_data pdata; struct i2c_client *i2c; struct snd_soc_jack *jack; struct delayed_work jack_detect_work; int sys_clk; + int clk_id; struct reg_default *index_cache; }; @@ -298,7 +300,6 @@ static int rt286_support_power_controls[] = { static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) { unsigned int val, buf; - int i; *hp = false; *mic = false; @@ -309,67 +310,44 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) if (*hp) { /* power on HV,VERF */ regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL1, 0x1001, 0x0); + RT286_DC_GAIN, 0x200, 0x200); + + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "HV"); + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "VREF"); /* power LDO1 */ - regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL2, 0x4, 0x4); - regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); - regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, + "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); - msleep(200); - i = 40; - while (((val & 0x0800) == 0) && (i > 0)) { - regmap_read(rt286->regmap, - RT286_CBJ_CTRL2, &val); - i--; - msleep(20); - } + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24); + msleep(50); - if (0x0400 == (val & 0x0700)) { - *mic = false; + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xfcc0, 0xd400); + msleep(300); + regmap_read(rt286->regmap, RT286_CBJ_CTRL2, &val); - regmap_write(rt286->regmap, - RT286_SET_MIC1, 0x20); - /* power off HV,VERF */ - regmap_update_bits(rt286->regmap, - RT286_POWER_CTRL1, 0x1001, 0x1001); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, 0xc000, 0x0000); - regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, 0x0030, 0x0000); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, 0xc000, 0x0000); - } else if ((0x0200 == (val & 0x0700)) || - (0x0100 == (val & 0x0700))) { + if (0x0070 == (val & 0x0070)) { *mic = true; - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, 0xc000, 0x8000); - regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, 0x0030, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, 0xc000, 0x8000); } else { - *mic = false; + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xfcc0, 0xe400); + msleep(300); + regmap_read(rt286->regmap, + RT286_CBJ_CTRL2, &val); + if (0x0070 == (val & 0x0070)) + *mic = true; + else + *mic = false; } - - regmap_update_bits(rt286->regmap, - RT286_MISC_CTRL1, - 0x0060, 0x0000); - } else { - regmap_update_bits(rt286->regmap, - RT286_MISC_CTRL1, - 0x0060, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL3, - 0xc000, 0x8000); regmap_update_bits(rt286->regmap, - RT286_CBJ_CTRL1, - 0x0030, 0x0020); - regmap_update_bits(rt286->regmap, - RT286_A_BIAS_CTRL2, - 0xc000, 0x8000); + RT286_DC_GAIN, 0x200, 0x0); + } else { *mic = false; + regmap_write(rt286->regmap, RT286_SET_MIC1, 0x20); } } else { regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf); @@ -378,6 +356,12 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) *mic = buf & 0x80000000; } + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "HV"); + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "VREF"); + if (!*hp) + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; } @@ -415,6 +399,17 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +static int is_mclk_mode(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(source->codec); + + if (rt286->clk_id == RT286_SCLK_S_MCLK) + return 1; + else + return 0; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -568,7 +563,84 @@ static int rt286_adc_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt286_vref_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0400, 0x0000); + mdelay(50); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_ldo2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x08); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT286_POWER_CTRL2, 0x38, 0x30); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_mic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY_S("HV", 1, RT286_POWER_CTRL1, + 12, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VREF", RT286_POWER_CTRL1, + 0, 1, rt286_vref_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT286_POWER_CTRL2, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("LDO2", 2, RT286_POWER_CTRL1, + 13, 1, rt286_ldo2_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("MCLK MODE", RT286_PLL_CTRL1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC1 Input Buffer", SND_SOC_NOPM, + 0, 0, rt286_mic1_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1 Pin"), SND_SOC_DAPM_INPUT("DMIC2 Pin"), @@ -642,6 +714,25 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"ADC 0", NULL, "MCLK MODE", is_mclk_mode}, + {"ADC 1", NULL, "MCLK MODE", is_mclk_mode}, + {"Front", NULL, "MCLK MODE", is_mclk_mode}, + {"Surround", NULL, "MCLK MODE", is_mclk_mode}, + + {"HP Power", NULL, "LDO1"}, + {"HP Power", NULL, "LDO2"}, + + {"MIC1", NULL, "LDO1"}, + {"MIC1", NULL, "LDO2"}, + {"MIC1", NULL, "HV"}, + {"MIC1", NULL, "VREF"}, + {"MIC1", NULL, "MIC1 Input Buffer"}, + + {"SPO", NULL, "LDO1"}, + {"SPO", NULL, "LDO2"}, + {"SPO", NULL, "HV"}, + {"SPO", NULL, "VREF"}, + {"DMIC1", NULL, "DMIC1 Pin"}, {"DMIC2", NULL, "DMIC2 Pin"}, {"DMIC1", NULL, "DMIC Receiver"}, @@ -880,6 +971,7 @@ static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, } rt286->sys_clk = freq; + rt286->clk_id = clk_id; return 0; } @@ -915,13 +1007,18 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: mdelay(10); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0400, 0x0400); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); + break; case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); snd_soc_update_bits(codec, - RT286_DC_GAIN, 0x200, 0x0); + RT286_CBJ_CTRL1, 0x0400, 0x0000); break; default: @@ -962,6 +1059,7 @@ static int rt286_probe(struct snd_soc_codec *codec) { struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + rt286->codec = codec; codec->dapm.bias_level = SND_SOC_BIAS_OFF; if (rt286->i2c->irq) { @@ -1152,7 +1250,6 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (!rt286->pdata.cbj_en) { regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); - regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); regmap_update_bits(rt286->regmap, RT286_CBJ_CTRL1, 0xf000, 0xb000); } else { @@ -1169,8 +1266,10 @@ static int rt286_i2c_probe(struct i2c_client *i2c, mdelay(10); - /*Power down LDO2*/ - regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + /*Power down LDO, VREF*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0xc, 0x0); + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL1, 0x1001, 0x1001); /*Set depop parameter*/ regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] [PATCH TINYCOMPRESS][RESEND] makefile.linux: allow env to override CROSS_COMPILE and CC
by Qais Yousef 04 Nov '14

04 Nov '14

allow env variables to override them without having to modify the makefile. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> --- I'm not sure why but I think the first send didn't reach the list, hence resending. makefile.linux | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/makefile.linux b/makefile.linux index 81782d0af461..c4dab6d3b7d5 100644 --- a/makefile.linux +++ b/makefile.linux @@ -9,8 +9,8 @@ LIBOBJ = ${LIBSRC:.c=.o} PLAYOBJ = ${PLAYSRC:.c=.o} RECOBJ = ${RECSRC:.c=.o} -CC = gcc -CROSS_COMPILE = +CROSS_COMPILE ?= +CC ?= ${CROSS_COMPILE}gcc CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress @@ -20,19 +20,19 @@ all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} + @${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} $(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} $(RECBIN): ${RECOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< - @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< + @${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} -- 2.1.0

1 0

[alsa-devel] [PATCH v2] snd-es18xx: Add GPO controls
by Ondrej Zary 04 Nov '14

04 Nov '14

Add GPO0 and GPO1 (General Purpose Outputs) controls to mixer. These can be used on some cards to control amplifier mute (seen in ES1868 datasheet) or additional onboard chips such as QX2130 QXpander processor. These GPOs are present on ES1868, ES1869, ES1887 and ES1888 chips. Tested on ES1868 with QX2130. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/isa/es18xx.c | 52 ++++++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 42 insertions(+), 10 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..0ead1c8 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -156,6 +156,7 @@ struct snd_es18xx { #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ #define ES18XX_CONTROL 0x0800 /* Has control ports */ +#define ES18XX_GPO_2BIT 0x1000 /* GPO0,1 controlled by PM port */ /* Power Management */ #define ES18XX_PM 0x07 @@ -1136,11 +1137,14 @@ static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg) return snd_es18xx_read(chip, reg); } -#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es18xx_info_single, \ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + .private_value = reg | (shift << 8) | (mask << 16) | (flags << 24) } + +#define ES18XX_FL_INVERT (1 << 0) +#define ES18XX_FL_PMPORT (1 << 1) static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1159,10 +1163,14 @@ static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; int val; - - val = snd_es18xx_reg_read(chip, reg); + + if (pm_port) + val = inb(chip->port + ES18XX_PM); + else + val = snd_es18xx_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; @@ -1175,7 +1183,8 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; unsigned char val; val = (ucontrol->value.integer.value[0] & mask); @@ -1183,6 +1192,15 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; mask <<= shift; val <<= shift; + if (pm_port) { + unsigned char cur = inb(chip->port + ES18XX_PM); + + if ((cur & mask) == val) + return 0; + outb((cur & ~mask) | val, chip->port + ES18XX_PM); + return 1; + } + return snd_es18xx_reg_bits(chip, reg, mask, val) != val; } @@ -1304,7 +1322,7 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, ES18XX_FL_INVERT), ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) @@ -1363,6 +1381,11 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; +static struct snd_kcontrol_new snd_es18xx_opt_gpo_2bit[] = { +ES18XX_SINGLE("GPO0 Switch", 0, ES18XX_PM, 0, 1, ES18XX_FL_PMPORT), +ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), +}; + static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1629,10 +1652,10 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_GPO_2BIT; break; case 0x1869: - chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV | ES18XX_GPO_2BIT; break; case 0x1878: chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; @@ -1642,7 +1665,7 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, break; case 0x1887: case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", @@ -1944,6 +1967,15 @@ static int snd_es18xx_mixer(struct snd_card *card) return err; } } + if (chip->caps & ES18XX_GPO_2BIT) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_gpo_2bit); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_gpo_2bit[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- Ondrej Zary

2 1

[alsa-devel] [Question about DPCM] dpcm_run_new_update races again xrun
by Qiao Zhou 04 Nov '14

04 Nov '14

Hi Mark, Liam I met one BE not-function issue when developing DPCM feature, and found dpcm_run_new_update is not protected well against xrun(interrupt context). Could you help to give some suggestions? The scenario is like this, taking audio playback for example: 1, FE1 <-> BE1, working well for some time. 2, disconnect BE1 and connect BE2(FE1 <-> BE2) 3, during FE1 connecting BE2, FE1 is still streaming data normally. Then an under-run happens. Below are the code sequence. soc_dpcm_runtime_update() { ... dpcm_connect_be() // connect FE1 & BE2 dpcm_run_new_update(fe, stream) { fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_BE dpcm_run_update_startup(fe, stream) { dpcm_be_dai_startup(fe, stream) _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ | // under-run happens (interrupt context) | | snd_pcm_stop(substream) { | | dpcm_fe_dai_trigger(STOP) { | | fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_FE | | // trigger stop FE1, BE2 | | fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_NO | | } | | } | |_ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _| dpcm_be_dai_hw_params(fe, stream) dpcm_be_dai_prepare(fe, stream) if(fe state == STOP) return 0; dpcm_be_dai_trigger(fe, ....) fe->dpcm.runtime_update = SND_SOC_DPCM_UPDATE_NO ... } } After xrun handler finishes, the FE runtime update staus is UPDATE_NO. Then the following dpcm_be_dai_hw_params & dpcm_be_dai_prepare skip related driver API excuting with this FE runtime status, and return 0 to end runtime-startup. When user APP(ALSA lib/TinyALSA) detects xrun, usually it will do the substream prepare and write data again. Due to BE dai has not been ready for hw_params, the BE dai can't work properly after substream prepare and trigger start. After that system has no sound and can't be recovered, maybe due to FE1 doesn't know what's going on inside BE2. The under-run is random, and under-run handler does what's necessary to be done, though trigger-stop a BE which has not been started yet is not good in my opinion. I did an experiment to avoid be udpate checking in dpcm_be_dai_hw_params by Commenting out "if(!snd_soc_dpcm_be_can_update()}". This change is verified OK since APP will do the prepare & trigger start to resume the under-run. The patch is also attached. >From current code, dpcm_be_dai_hw_params is only callded by fe_hw_params & this runtime startup. This be update checking might be redundant or unnecessary in current code context. Could you help to give some suggestions? Please help to correct me if anything wrong. Thanks in advance. --- sound/soc/soc-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..dfeb1e7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1697,10 +1697,6 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); - /* is this op for this BE ? */ - if (!snd_soc_dpcm_be_can_update(fe, be, stream)) - continue; - /* only allow hw_params() if no connected FEs are running */ if (!snd_soc_dpcm_can_be_params(fe, be, stream)) continue; -- 1.7.9.5 Best Regards Qiao

4 8

[alsa-devel] [PATCHv4] ASoC: simple-card: binding: update binding to support the new style.
by Xiubo Li 04 Nov '14

04 Nov '14

This patch will split the DT binding into old style and new style, the new style will be easier to add many DAI links from old single DAI link DTs. This patch will maintian compatibility with the old DTs. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changes in v4: - Follow Jyri's advice. - Fix some small words or sentences errors. - Fix discription of the commit commnet. - Apply against the current code. - Add Notes. .../devicetree/bindings/sound/simple-card.txt | 215 ++++++++++++++------- 1 file changed, 148 insertions(+), 67 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c2e9841..5fe0796 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -1,15 +1,19 @@ -Simple-Card: +Device-Tree bindings for Simple Card Simple-Card specifies audio DAI connections of SoC <-> codec. -Required properties: +=== Top level's properties and subnodes === +*** Required properties *** - compatible : "simple-audio-card" -Optional properties: - +*** Optional properties *** - simple-audio-card,name : User specified audio sound card name, one string property. +- simple-audio-card,format : CPU/CODEC common audio format. + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" + (This is used for single DAI link & old style.) - simple-audio-card,widgets : Please refer to widgets.txt. - simple-audio-card,routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the @@ -17,63 +21,99 @@ Optional properties: source. - simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec mclk. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section bellow. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - +- simple-audio-card,frame-master : Indicates DAI link frame master. One phandle to a + CODEC DAI node. + (This is used for single DAI link & old style.) +- simple-audio-card,bitclock-master : Indicates DAI link bit clock master. One phandle to + a CODEC DAI node. + (This is used for single DAI link & old style.) + +*** Optional subnodes *** +- simple-audio-card,dai-link : Container for DAI link level properties and the CPU + and CODEC nodes. This container may be omitted + when the card has only one DAI link and using the old + style. See the examples and the section bellow. +- simple-audio-card,cpu : CPU DAI node. + (This is used for single DAI link & old style.) +- simple-audio-card,codec : CODEC DAI node. + (This is used for single DAI link & old style.) + +=== DAI link node's properties and its subnodes === + +*** Required subnodes *** +- cpu : CPU DAI node. +- codec : CODEC DAI node. + +*** Optional properties *** - format : CPU/CODEC common audio format. "i2s", "right_j", "left_j" , "dsp_a" "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - +- frame-master : Indicates DAI link frame master. One phandle to a + CODEC DAI node. + (This is One boolean property for old style.) +- bitclock-master : Indicates DAI link bit clock master. One phandle to + a CODEC DAI node. + (This is one boolean property for old style.) + +For backward compatibility, the frame-master and bitclock-master +can be used as boolean properties in CPU/CODEC DAI node indicating +that the CODEC DAI is the DAI link frame master and bit clock master. +In this case there should be no DAI link level nodes or boolean +properties of them. + +For all the properties in DAI link level will be applied to both +CPU and CODEC DAIs. + +=== CPU/CODEC DAI node's properties and its subnodes === + +*** Required properties *** +- sound-dai : One phandle and port of CPU/CODEC DAI device node. + +*** Optional properties *** +- bitclock-inversion : Boolean property. Add this if the DAI device uses bit + clock inversion. +- frame-inversion : Boolean property. Add this if the DAI device uses frame + clock inversion. +- frame-master : Boolean property. Add this if the CODEC DAI device is + frame master. + (This is used for single DAI link & old style.) +- bitclock-master : Boolean property. Add this if the CODEC DAI device is + bit clock master. + (This is used for single DAI link & old style.) - dai-tdm-slot-num : Please refer to tdm-slot.txt. - dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) +- clocks / system-clock-frequency : Specify CPU/CODEC DAI's clock if needed. It can + be specified via "clocks" if system has clock node + (= common clock), or "system-clock-frequency"(if system + doens't support common clock) + +=== Notes === +*** DAI hardware clock masters *** +For [prefix]frame-master and [prefix]bitclock-master, which prefix +is "simple-audio-card,", are both wrt the CODEC DAI. If the CODEC DAI +is bit clock and frame master then the CPU DAI will be bit clock and +frame slave, and vice versa. + +=== Examples === +*** CPU & CODEC DAI DT nodes *** +&i2c0 { + ak4648: ak4648@12 { + #sound-dai-cells = <0>; + compatible = "asahi-kasei,ak4648"; + reg = <0x12>; + }; +}; -Example 1 - single DAI link: +sh_fsi2: sh_fsi2@ec230000 { + #sound-dai-cells = <1>; + compatible = "renesas,sh_fsi2"; + reg = <0xec230000 0x400>; + interrupt-parent = <&gic>; + interrupts = <0 146 0x4>; +}; +Example 1 - single DAI link & old style: +bitclock-master and frame-master as phandles. sound { compatible = "simple-audio-card"; simple-audio-card,name = "VF610-Tower-Sound-Card"; @@ -91,6 +131,7 @@ sound { simple-audio-card,cpu { sound-dai = <&sh_fsi2 0>; + bitclock-inversion; }; dailink0_master: simple-audio-card,codec { @@ -99,24 +140,64 @@ sound { }; }; -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; +Example 2 - single DAI link & old style: +bitclock-master and frame-master as boolean properties. +sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; }; -}; -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; + simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + bitclock-master; + frame-master; + bitclock-inversion; + }; }; -Example 2 - many DAI links: +Example 3 - single DAI link & new style: +sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,dai-link { + format = "i2s"; + bitclock-master = <&dailink0_master>; + frame-master = <&dailink0_master>; + cpu { + sound-dai = <&sh_fsi2 0>; + frame-inversion; + }; + dailink0_master: codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + frame-inversion; + }; + }; +}; +Example 4 - many DAI links: sound { compatible = "simple-audio-card"; simple-audio-card,name = "Cubox Audio"; -- 2.1.0.27.g96db324

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic23: depends on SPI
by Sudip Mukherjee 04 Nov '14

04 Nov '14

Fix build errors when CONFIG_SPI is not enabled by making the driver depend on SPI. ERROR: "spi_setup" [sound/soc/codecs/snd-soc-tlv320aic23-spi.ko] undefined! ERROR: "devm_regmap_init_spi" [sound/soc/codecs/snd-soc-tlv320aic23-spi.ko] undefined! ERROR: "spi_register_driver" [sound/soc/codecs/snd-soc-tlv320aic23-spi.ko] undefined! Reported-by: Jim Davis <jim.epost(a)gmail.com> Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- reference of mail by Jim Davis at https://lkml.org/lkml/2014/11/3/551 sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6801b38..9a8ea56 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -603,6 +603,7 @@ config SND_SOC_TLV320AIC23_I2C config SND_SOC_TLV320AIC23_SPI tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" + depends on SPI select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- 1.8.1.2

2 1

[alsa-devel] [PATCH v2 1/2] ALSA: echoaudio: add reference of struct echoaudio
by Sudip Mukherjee 03 Nov '14

03 Nov '14

added reference of struct echoaudio to free_firmware function. this structure will be later used to get a reference of the card when converting snd_printk to dev_* in the next patch of the series. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- same as v1. sound/pci/echoaudio/echoaudio.c | 3 ++- sound/pci/echoaudio/echoaudio.h | 3 ++- sound/pci/echoaudio/echoaudio_dsp.c | 10 +++++----- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d82321f..db1b247 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -69,7 +69,8 @@ static int get_firmware(const struct firmware **fw_entry, -static void free_firmware(const struct firmware *fw_entry) +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip) { #ifdef CONFIG_PM_SLEEP DE_ACT(("firmware not released (kept in cache)\n")); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index b86b88d..a4f112a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -468,7 +468,8 @@ static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, struct echoaudio *chip, const short fw_index); -static void free_firmware(const struct firmware *fw_entry); +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 5a6a217..977b2bd 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) } DE_INIT(("ASIC loaded\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; la_error: DE_INIT(("failed on write_dsp\n")); - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -317,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip) } DE_INIT(("Resident loader successfully installed\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; irl_error: - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -491,7 +491,7 @@ static int load_firmware(struct echoaudio *chip) if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); - free_firmware(fw); + free_firmware(fw, chip); if (err < 0) return err; -- 1.8.1.2

3 6

[alsa-devel] [PATCH] snd_user_file: fix memory leak
by Andreas Schwab 03 Nov '14

03 Nov '14

Call wordfree if and only if wordfree returns zero or WRDE_NOSPACE Signed-off-by: Andreas Schwab <schwab(a)suse.de> --- src/userfile.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/userfile.c b/src/userfile.c index 3a73836..72779da 100644 --- a/src/userfile.c +++ b/src/userfile.c @@ -44,19 +44,20 @@ int snd_user_file(const char *file, char **result) err = wordexp(file, &we, WRDE_NOCMD); switch (err) { case WRDE_NOSPACE: + wordfree(&we); return -ENOMEM; case 0: if (we.we_wordc == 1) break; + wordfree(&we); /* fall thru */ default: - wordfree(&we); return -EINVAL; } *result = strdup(we.we_wordv[0]); + wordfree(&we); if (*result == NULL) return -ENOMEM; - wordfree(&we); return 0; } -- 2.1.3 -- Andreas Schwab, SUSE Labs, schwab(a)suse.de GPG Key fingerprint = 0196 BAD8 1CE9 1970 F4BE 1748 E4D4 88E3 0EEA B9D7 "And now for something completely different."

2 1

[alsa-devel] [PATCH] USB-Audio: Add Scarlett 2i4 USB to S/PDIF blacklist
by Panu Matilainen 03 Nov '14

03 Nov '14

Like Scarlett 2i2, the 2i4 does not have any S/PDIF connections. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 77a48b9..a6754d9 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -48,6 +48,7 @@ USB-Audio.pcm.iec958_device { "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 "Scarlett 2i2 USB" 999 + "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 "USB Device 0x46d:0x992" 999 } -- 1.9.3

2 1

[alsa-devel] [PATCH] ASoC: Intel: Correct a macro for FW message
by Jie Yang 03 Nov '14

03 Nov '14

For the broadwell official released FW(Since 8.4.1.43), the macro SST_HSW_NO_CHANNELS is changed and fixed to 4, so here change it to 4. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index afd0ae1..387511f 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -21,7 +21,7 @@ #include <linux/kernel.h> #include <linux/platform_device.h> -#define SST_HSW_NO_CHANNELS 2 +#define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 #define SST_HSW_DX_CONTEXT_SIZE (640 * 1024) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Add Merrifield machine driver
by Vinod Koul 03 Nov '14

03 Nov '14

This patch adds machine driver support for Merrifield platform. Patch has dependency on Merrifield DSP driver. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- this is standalone patch with no depends on other code posted sound/soc/intel/Kconfig | 15 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/merr_dpcm_wm8958.c | 430 ++++++++++++++++++++++++++++++++++++ 3 files changed, 447 insertions(+), 0 deletions(-) create mode 100644 sound/soc/intel/merr_dpcm_wm8958.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..16d1801 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -76,3 +76,18 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_MRFLD_WM8958_MACH + tristate "SOC Machine Audio driver for Intel Merrifield MID platform" + depends on X86 + select SND_SOC_WM8994 + select MFD_CORE + select MFD_WM8994 + select REGULATOR_WM8994 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC + help + This adds support for ASoC machine driver for Intel(R) MID Merrifield platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..9d4b3d4 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -26,6 +26,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o +snd-merr-dpcm-wm8958-objs := merr_dpcm_wm8958.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -34,3 +35,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ +obj-$(CONFIG_SND_SOC_INTEL_MRFLD_WM8958_MACH) += snd-merr-dpcm-wm8958.o diff --git a/sound/soc/intel/merr_dpcm_wm8958.c b/sound/soc/intel/merr_dpcm_wm8958.c new file mode 100644 index 0000000..4ac6a72 --- /dev/null +++ b/sound/soc/intel/merr_dpcm_wm8958.c @@ -0,0 +1,430 @@ +/* + * merr_dpcm_wm8958.c - ASoc DPCM Machine driver for Intel Merrfield MID platform + * + * Copyright (C) 2013-14 Intel Corp + * Author: Vinod Koul <vinod.koul(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/async.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <linux/input.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include <linux/mfd/wm8994/pdata.h> +#include "../codecs/wm8994.h" +#include "sst-atom-controls.h" + +/* Codec PLL output clk rate */ +#define CODEC_SYSCLK_RATE 24576000 +/* Input clock to codec at MCLK1 PIN */ +#define CODEC_IN_MCLK1_RATE 19200000 +/* Input clock to codec at MCLK2 PIN */ +#define CODEC_IN_MCLK2_RATE 32768 +/* define to select between MCLK1 and MCLK2 input to codec as its clock */ +#define CODEC_IN_MCLK1 1 +#define CODEC_IN_MCLK2 2 +#define CODEC_BE 1 + +struct mrfld_8958_mc_private { + struct snd_soc_jack jack; + int jack_retry; +}; + +/* + * Function to switch the input clock for codec, When audio is in + * progress input clock to codec will be through MCLK1 which is 19.2MHz + * while in off state input clock to codec will be through 32KHz through + * MCLK2 + * card : Sound card structure + * src : Input clock source to codec + */ +static int mrfld_8958_set_codec_clk(struct snd_soc_card *card, int src) +{ + int ret; + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + switch (src) { + case CODEC_IN_MCLK1: + /* Turn ON the PLL to generate required sysclk rate + * from MCLK1 + */ + ret = snd_soc_dai_set_pll(aif1_dai, + WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + CODEC_IN_MCLK1_RATE, CODEC_SYSCLK_RATE); + if (ret < 0) { + dev_err(card->dev, "Failed to start FLL: %d\n", ret); + return ret; + } + /* Switch to MCLK1 input */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_FLL1, + CODEC_SYSCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Failed to set codec sysclk configuration %d\n", + ret); + return ret; + } + break; + case CODEC_IN_MCLK2: + /* Switch to MCLK2 */ + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, + 32768, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Failed to switch to MCLK2: %d\n", ret); + return ret; + } + /* Turn off PLL for MCLK1 */ + ret = snd_soc_dai_set_pll(aif1_dai, WM8994_FLL1, 0, 0, 0); + if (ret < 0) { + dev_err(card->dev, "Failed to stop the FLL: %d", ret); + return ret; + } + break; + default: + return -EINVAL; + } + return 0; +} + +static int mrfld_wm8958_set_clk(struct snd_soc_dai *codec_dai) +{ + int ret = 0; + struct snd_soc_card *card = codec_dai->card; + + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf0, 0xf0, 4, 24); + if (ret < 0) { + dev_err(card->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK1); + + return ret; +} + +static int mrfld_8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + return mrfld_wm8958_set_clk(codec_dai); +} + +static const struct snd_soc_pcm_stream mrfld_wm8958_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int merr_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int mrfld_8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + + ret = mrfld_wm8958_set_clk(aif1_dai); + break; + default: + break; + } + return ret; +} + +static int mrfld_8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + int ret = 0; + /* "wm8994-aif1" */ + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + + if (!aif1_dai) + return -ENODEV; + + if (dapm->dev != aif1_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + /* + * We are in standby down so */ + /* Switch to 32KHz MCLK2 input clock for codec + */ + ret = mrfld_8958_set_codec_clk(card, CODEC_IN_MCLK2); + break; + default: + break; + } + if (&card->dapm == dapm) + card->dapm.bias_level = level; + return ret; +} + +static const struct snd_soc_dapm_widget widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_route map[] = { + { "Headphones", NULL, "HPOUT1L" }, + { "Headphones", NULL, "HPOUT1R" }, + + /* + * This machine uses 2 DMICs, other configs may use more so change + * below accordingly + */ + { "DMIC1DAT", NULL, "DMIC" }, + { "DMIC2DAT", NULL, "DMIC" }, + + /* + * MICBIAS2 is connected as Bias for AMIC so we link it + * here. Also AMIC wires up to IN1LP pin. + * DMIC is externally connected to 1.8V rail, so no link rqd. + */ + { "AMIC", NULL, "MICBIAS2" }, + { "IN1LP", NULL, "AMIC" }, + + /* dsp map link the SWM outs to codec AIF */ + { "AIF1 Playback", NULL, "ssp2 Tx"}, + { "ssp2 Tx", NULL, "codec_out0"}, + { "ssp2 Tx", NULL, "codec_out1"}, + { "codec_in0", NULL, "ssp2 Rx" }, + { "codec_in1", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture"}, + +}; + +static const struct wm8958_micd_rate micdet_rates[] = { + { 32768, true, 1, 4 }, + { 32768, false, 1, 1 }, + { 44100 * 256, true, 7, 10 }, + { 44100 * 256, false, 7, 10 }, +}; + +static int mrfld_8958_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + struct mrfld_8958_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *aif1_dai = card->rtd[CODEC_BE].codec_dai; + struct snd_soc_codec *codec = aif1_dai->codec; + + if (!aif1_dai) + return -ENODEV; + + ret = snd_soc_dai_set_tdm_slot(aif1_dai, 0xf0, 0xf0, 4, 24); + if (ret < 0) { + dev_err(card->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + if (!codec) { + dev_err(card->dev, "%s: we didnt find the codec pointer!\n", __func__); + return 0; + } + + ctx->jack_retry = 0; + ret = snd_soc_jack_new(codec, "Intel MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1, + &ctx->jack); + if (ret) { + dev_err(card->dev, "jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_1, KEY_MEDIA); + snd_jack_set_key(ctx->jack.jack, SND_JACK_BTN_0, KEY_MEDIA); + + wm8958_mic_detect(codec, &ctx->jack, NULL, NULL, NULL, NULL); + + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; +static int mrfld_8958_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops mrfld_8958_ops = { + .startup = mrfld_8958_startup, +}; + +static struct snd_soc_ops mrfld_8958_be_ssp2_ops = { + .hw_params = mrfld_8958_hw_params, +}; + +struct snd_soc_dai_link mrfld_8958_msic_dailink[] = { + [MERR_DPCM_AUDIO] = { + .name = "Merrifield Audio Port", + .stream_name = "Merrifield Audio", + .cpu_dai_name = "Headset-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "sst-platform", + .init = mrfld_8958_init, + .ignore_suspend = 1, + .dynamic = 1, + .ops = &mrfld_8958_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Merrifield Compress Port", + .stream_name = "Merrifield Compress", + .platform_name = "sst-platform", + .cpu_dai_name = "Compress-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .init = mrfld_8958_init, + }, + + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-platform", + .no_pcm = 1, + .codec_dai_name = "wm8994-aif1", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = merr_codec_fixup, + .ignore_suspend = 1, + .ops = &mrfld_8958_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mrfld = { + .name = "wm8958-audio", + .dai_link = mrfld_8958_msic_dailink, + .num_links = ARRAY_SIZE(mrfld_8958_msic_dailink), + .set_bias_level = mrfld_8958_set_bias_level, + .set_bias_level_post = mrfld_8958_set_bias_level_post, + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = map, + .num_dapm_routes = ARRAY_SIZE(map), + .fully_routed = true, +}; + +static int snd_mrfld_8958_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct mrfld_8958_mc_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (!drv) + return -ENOMEM; + + /* register the soc card */ + snd_soc_card_mrfld.dev = &pdev->dev; + snd_soc_card_set_drvdata(&snd_soc_card_mrfld, drv); + ret_val = snd_soc_register_card(&snd_soc_card_mrfld); + if (ret_val) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret_val); + goto unalloc; + } + platform_set_drvdata(pdev, &snd_soc_card_mrfld); + return ret_val; + +unalloc: + return ret_val; +} + +static int snd_mrfld_8958_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_mrfld_8958_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "mrfld_wm8958", + }, + .probe = snd_mrfld_8958_mc_probe, + .remove = snd_mrfld_8958_mc_remove, +}; + +module_platform_driver(snd_mrfld_8958_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Merrifield MID Machine driver"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mrfld_wm8958"); -- 1.7.0.4

2 4

[alsa-devel] Query - PyAlsa Docs
by Chirag Ajmera 03 Nov '14

03 Nov '14

Dear, Where can I get documentation for pyalsa bindings ? The test files (eg cardtest1.py) included in the package fail with the below error ImportError: No module named alsacard import pyalsa - this package is found and loaded Kindly guide Thank you Best, Chirag A.

1 0

[alsa-devel] [PATCH 1/1] PCI: EMU10K1: Deletion of unnecessary checks before three function calls
by SF Markus Elfring 03 Nov '14

03 Nov '14

The functions kfree(), release_firmware() and snd_util_memhdr_free() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/emu10k1/emu10k1_main.c | 9 +++------ sound/pci/emu10k1/emufx.c | 3 +-- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2292697..b4458a6 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1289,10 +1289,8 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) } if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); - if (emu->firmware) - release_firmware(emu->firmware); - if (emu->dock_fw) - release_firmware(emu->dock_fw); + release_firmware(emu->firmware); + release_firmware(emu->dock_fw); if (emu->irq >= 0) free_irq(emu->irq, emu); /* remove reserved page */ @@ -1301,8 +1299,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) (struct snd_util_memblk *)emu->reserved_page); emu->reserved_page = NULL; } - if (emu->memhdr) - snd_util_memhdr_free(emu->memhdr); + snd_util_memhdr_free(emu->memhdr); if (emu->silent_page.area) snd_dma_free_pages(&emu->silent_page); if (emu->ptb_pages.area) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 745f062..eb5c0ab 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -777,8 +777,7 @@ static void snd_emu10k1_ctl_private_free(struct snd_kcontrol *kctl) kctl->private_value = 0; list_del(&ctl->list); kfree(ctl); - if (kctl->tlv.p) - kfree(kctl->tlv.p); + kfree(kctl->tlv.p); } static int snd_emu10k1_add_controls(struct snd_emu10k1 *emu, -- 2.1.3

2 3

[alsa-devel] [PATCH 0/3] Trivial usb-audio cleanup patches
by Takashi Iwai 03 Nov '14

03 Nov '14

Hi, here are a few trivial cleanup patches for usb-audio driver. [PATCH 1/3] ALSA: usb-audio: Flatten probe and disconnect functions [PATCH 2/3] ALSA: usb-audio: Pass direct struct pointer instead of [PATCH 3/3] ALSA: usb-audio: Use strim() instead of open code Takashi

1 3

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex.
by Peter Rosin 03 Nov '14

03 Nov '14

>From 86be84c4de4e7b21cfda9656a02a902c543210af Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Date: Wed, 22 Oct 2014 16:45:29 +0200 Subject: [PATCH v2] ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. The CMR divider register is shared by playback and capture. The SSC driver therefore tries to enforce rules so that the needed register content do not conflict during simultaneous playback/capture. However, the implementation also prevents changing the register content in half-duplex scenarios, which is needed when using the OSS API. Thus, only lock the divider if there is a stream in the other direction. Fixes the below program to not fail with the atmel ssc dai in master mode. #include <sys/ioctl.h> #include <unistd.h> #include <fcntl.h> #include <sys/soundcard.h> int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22025; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index de433cfd..9ae8475 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, * transmit and receive, so if a value has already * been set, it must match this value. */ - if (ssc_p->cmr_div == 0) + if (ssc_p->dir_mask != + (SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE)) + ssc_p->cmr_div = div; + else if (ssc_p->cmr_div == 0) ssc_p->cmr_div = div; else if (div != ssc_p->cmr_div) -- 1.7.10.4

3 9

[alsa-devel] [PATCH] ASoC: simple-card: add "invert" property for detect GPIOs
by Jianqun 03 Nov '14

03 Nov '14

Since hardware may invert detect GPIO of headphone or mic, add one property to support software invert. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/generic/simple-card.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d1b7293..f1bb537 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -29,7 +29,9 @@ struct simple_card_data { } *dai_props; unsigned int mclk_fs; int gpio_hp_det; + int gpio_hp_det_invert; int gpio_mic_det; + int gpio_mic_det_invert; struct snd_soc_dai_link dai_link[]; /* dynamically allocated */ }; @@ -148,6 +150,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) simple_card_hp_jack_pins); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; + simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; snd_soc_jack_add_gpios(&simple_card_hp_jack, 1, &simple_card_hp_jack_gpio); } @@ -159,6 +162,7 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(simple_card_mic_jack_pins), simple_card_mic_jack_pins); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; + simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); } @@ -381,6 +385,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + enum of_gpio_flags flags; u32 val; int ret; @@ -436,13 +441,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } - priv->gpio_hp_det = of_get_named_gpio(node, - "simple-audio-card,hp-det-gpio", 0); + priv->gpio_hp_det = of_get_named_gpio_flags(node, + "simple-audio-card,hp-det-gpio", 0, &flags); + priv->gpio_hp_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; - priv->gpio_mic_det = of_get_named_gpio(node, - "simple-audio-card,mic-det-gpio", 0); + priv->gpio_mic_det = of_get_named_gpio_flags(node, + "simple-audio-card,mic-det-gpio", 0, &flags); + priv->gpio_mic_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); if (priv->gpio_mic_det == -EPROBE_DEFER) return -EPROBE_DEFER; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: Remove snd_soc_platform_driver suspend/resume callbacks
by Lars-Peter Clausen 03 Nov '14

03 Nov '14

Those are unused and new drivers should use device driver suspend/resume. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 ---- sound/soc/soc-core.c | 10 ---------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ad47e96..edbb07b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -857,8 +857,6 @@ struct snd_soc_platform_driver { int (*probe)(struct snd_soc_platform *); int (*remove)(struct snd_soc_platform *); - int (*suspend)(struct snd_soc_dai *dai); - int (*resume)(struct snd_soc_dai *dai); struct snd_soc_component_driver component_driver; /* pcm creation and destruction */ @@ -891,8 +889,6 @@ struct snd_soc_platform { struct device *dev; const struct snd_soc_platform_driver *driver; - unsigned int suspended:1; /* platform is suspended */ - struct list_head list; struct snd_soc_component component; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dd8234d..00e944e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -592,17 +592,12 @@ int snd_soc_suspend(struct device *dev) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control) cpu_dai->driver->suspend(cpu_dai); - if (platform->driver->suspend && !platform->suspended) { - platform->driver->suspend(cpu_dai); - platform->suspended = 1; - } } /* close any waiting streams and save state */ @@ -775,17 +770,12 @@ static void soc_resume_deferred(struct work_struct *work) for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - struct snd_soc_platform *platform = card->rtd[i].platform; if (card->rtd[i].dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control) cpu_dai->driver->resume(cpu_dai); - if (platform->driver->resume && platform->suspended) { - platform->driver->resume(cpu_dai); - platform->suspended = 0; - } } if (card->resume_post) -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: rockchip-i2s: add rockchip-max98090 driver support
by Jianqun 03 Nov '14

03 Nov '14

This patch is to add support for rockchip board with max98090 codec. It also needs sound node in dt file. Tested on rockchip board based on rk3288. Signed-off-by: Jianqun <jay.xu(a)rock-chips.com> --- sound/soc/rockchip/Kconfig | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 78fc159..482cbba 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -9,3 +9,16 @@ config SND_SOC_ROCKCHIP config SND_SOC_ROCKCHIP_I2S tristate + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP + help + Say Y or M if you want to add support for I2S driver for Rockchip + I2S device. + +config SND_SOC_ROCKCHIP_MAX98090 + tristate "ASoC support Rockchip board with max98090" + depends on I2C && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98090 + help + Say Y or M here if you want to add support for Rockchio board using + a max98090 codec. -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: tlv320aic23: add dependencies on I2C/SPI_MASTER
by Max Filippov 03 Nov '14

03 Nov '14

This fixes build errors in configurations with I2C/SPI master disabled. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7dad805..b0bcd0d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -553,10 +553,12 @@ config SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_I2C tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C" + depends on I2C select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_SPI tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" + depends on SPI_MASTER select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- 1.8.1.4

2 1

[alsa-devel] [PATCH] [RESEND] snd-es18xx: Add GPO controls
by Ondrej Zary 03 Nov '14

03 Nov '14

Add GPO0 and GPO1 (General Purpose Outputs) controls to mixer. These can be used on some cards to control amplifier mute (seen in ES1868 datasheet) or additional onboard chips such as QX2130 QXpander processor. These GPOs are present on ES1868, ES1869, ES1887 and ES1888 chips. Tested on ES1868 with QX2130. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/isa/es18xx.c | 49 +++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 39 insertions(+), 10 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 6faaac6..8966ed2 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -156,6 +156,7 @@ struct snd_es18xx { #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ #define ES18XX_CONTROL 0x0800 /* Has control ports */ +#define ES18XX_GPO_2BIT 0x1000 /* GPO0,1 controlled by PM port */ /* Power Management */ #define ES18XX_PM 0x07 @@ -1136,11 +1137,14 @@ static int snd_es18xx_reg_read(struct snd_es18xx *chip, unsigned char reg) return snd_es18xx_read(chip, reg); } -#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define ES18XX_SINGLE(xname, xindex, reg, shift, mask, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_es18xx_info_single, \ .get = snd_es18xx_get_single, .put = snd_es18xx_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + .private_value = reg | (shift << 8) | (mask << 16) | (flags << 24) } + +#define ES18XX_FL_INVERT (1 << 0) +#define ES18XX_FL_PMPORT (1 << 1) static int snd_es18xx_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1159,10 +1163,14 @@ static int snd_es18xx_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; int val; - - val = snd_es18xx_reg_read(chip, reg); + + if (pm_port) + val = inb(chip->port + ES18XX_PM); + else + val = snd_es18xx_reg_read(chip, reg); ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; @@ -1175,7 +1183,8 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 24) & ES18XX_FL_INVERT; + int pm_port = (kcontrol->private_value >> 24) & ES18XX_FL_PMPORT; unsigned char val; val = (ucontrol->value.integer.value[0] & mask); @@ -1183,6 +1192,12 @@ static int snd_es18xx_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; mask <<= shift; val <<= shift; + if (pm_port) { + val = (inb(chip->port + ES18XX_PM) & ~mask) | val; + outb(val, chip->port + ES18XX_PM); + return 0; + } + return snd_es18xx_reg_bits(chip, reg, mask, val) != val; } @@ -1304,7 +1319,7 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, ES18XX_FL_INVERT), ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) @@ -1363,6 +1378,11 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; +static struct snd_kcontrol_new snd_es18xx_opt_gpo_2bit[] = { +ES18XX_SINGLE("GPO0 Switch", 0, ES18XX_PM, 0, 1, ES18XX_FL_PMPORT), +ES18XX_SINGLE("GPO1 Switch", 0, ES18XX_PM, 1, 1, ES18XX_FL_PMPORT), +}; + static int snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1629,10 +1649,10 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_GPO_2BIT; break; case 0x1869: - chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV | ES18XX_GPO_2BIT; break; case 0x1878: chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; @@ -1642,7 +1662,7 @@ static int snd_es18xx_probe(struct snd_es18xx *chip, break; case 0x1887: case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", @@ -1944,6 +1964,15 @@ static int snd_es18xx_mixer(struct snd_card *card) return err; } } + if (chip->caps & ES18XX_GPO_2BIT) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_gpo_2bit); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_gpo_2bit[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- Ondrej Zary

2 1

[alsa-devel] [PATCH 1/2] ALSA: echoaudio: add reference of struct echoaudio
by Sudip Mukherjee 03 Nov '14

03 Nov '14

added reference of struct echoaudio to free_firmware function. this structure will be later used to get a reference of the card when converting snd_printk to dev_* in the next patch of the series. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/echoaudio/echoaudio.c | 3 ++- sound/pci/echoaudio/echoaudio.h | 3 ++- sound/pci/echoaudio/echoaudio_dsp.c | 10 +++++----- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d82321f..db1b247 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -69,7 +69,8 @@ static int get_firmware(const struct firmware **fw_entry, -static void free_firmware(const struct firmware *fw_entry) +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip) { #ifdef CONFIG_PM_SLEEP DE_ACT(("firmware not released (kept in cache)\n")); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index b86b88d..a4f112a 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -468,7 +468,8 @@ static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, struct echoaudio *chip, const short fw_index); -static void free_firmware(const struct firmware *fw_entry); +static void free_firmware(const struct firmware *fw_entry, + struct echoaudio *chip); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 5a6a217..977b2bd 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -206,12 +206,12 @@ static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) } DE_INIT(("ASIC loaded\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; la_error: DE_INIT(("failed on write_dsp\n")); - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -317,11 +317,11 @@ static int install_resident_loader(struct echoaudio *chip) } DE_INIT(("Resident loader successfully installed\n")); - free_firmware(fw); + free_firmware(fw, chip); return 0; irl_error: - free_firmware(fw); + free_firmware(fw, chip); return -EIO; } @@ -491,7 +491,7 @@ static int load_firmware(struct echoaudio *chip) if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); - free_firmware(fw); + free_firmware(fw, chip); if (err < 0) return err; -- 1.8.1.2

2 2

[alsa-devel] [PATCH 0/3] Kill the rest snd_print*() usages
by Takashi Iwai 03 Nov '14

03 Nov '14

Hi, it's a small series to get rid of snd_print*() usages in a few PCI sound drivers. There remain still a few, I know, but this is just a matter of cleanup, so let them be there... [PATCH 1/3] ALSA: ctxfi: Kill the rest snd_print*() [PATCH 2/3] ALSA: pcxhr: Kill the rest snd_print*() [PATCH 3/3] ALSA: au88x0: Kill the rest snd_print*() Takashi

3 8

Re: [alsa-devel] ASoC: Intel: mrfld: add DSP core controls
by Dan Carpenter 03 Nov '14

03 Nov '14

Hello Vinod Koul, The patch 24c8d14192cc: "ASoC: Intel: mrfld: add DSP core controls" from Oct 15, 2014, leads to the following static checker warning: sound/soc/intel/sst-atom-controls.c:1354 sst_map_modules_to_pipe() warn: variable dereferenced before check 'platform' (see line 1353) sound/soc/intel/sst-atom-controls.c 1348 static int sst_map_modules_to_pipe(struct snd_soc_platform *platform) 1349 { 1350 struct snd_soc_dapm_widget *w; 1351 int ret = 0; 1352 1353 list_for_each_entry(w, &platform->component.card->widgets, list) { ^^^^^^^^ New dereference. 1354 if (platform && is_sst_dapm_widget(w) && (w->priv)) { ^^^^^^^^ New check. 1355 struct sst_ids *ids = w->priv; 1356 1357 dev_dbg(platform->dev, "widget type=%d name=%s\n", 1358 w->id, w->name); 1359 INIT_LIST_HEAD(&ids->algo_list); 1360 INIT_LIST_HEAD(&ids->gain_list); 1361 ret = sst_fill_widget_module_info(w, platform); 1362 1363 if (ret < 0) 1364 return ret; 1365 1366 /* fill linked widgets */ 1367 if (ids->parent_wname != NULL) 1368 sst_fill_linked_widgets(platform, ids); 1369 } 1370 } 1371 return 0; 1372 } regards, dan carpenter

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: add compressed support
by Vinod Koul 03 Nov '14

03 Nov '14

This add compressed support in sst ipc driver. Along with it two minor fixes on driver as well Fang, Yang A (1): ASoC: Intel: use correct firmware name Subhransu S. Prusty (1): ASoC: mfld-compress: implement .power callback Vinod Koul (2): ASoC: Intel: sst: load firmware using async callback ASoC: Intel: sst - add compressed ops handling sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform.h | 1 + sound/soc/intel/sst/sst.c | 15 ++ sound/soc/intel/sst/sst_drv_interface.c | 285 ++++++++++++++++++++++++++ sound/soc/intel/sst/sst_loader.c | 5 +- 5 files changed, 309 insertions(+), 5 deletions(-)

2 9

[alsa-devel] [PATCH] audio: sai: Add Power Management support
by Alison Wang 03 Nov '14

03 Nov '14

This patch adds Power Management support for SAI. Activate regmap cache with REGCACHE_RBTREE, and use regmap cache code to save and restore registers in suspend and resume. The Transmit Control Register (TCSR) and Receive Control Register(RCSR) should be volatile registers. Signed-off-by: Alison Wang <alison.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 ++++++++++++++++++++++++++++++++- 1 file changed, 32 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7eeb1dd..c7dd953 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -509,9 +509,11 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: case FSL_SAI_TFR: - case FSL_SAI_RFR: case FSL_SAI_TDR: + case FSL_SAI_RCSR: + case FSL_SAI_RFR: case FSL_SAI_RDR: return true; default: @@ -553,6 +555,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_RBTREE, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -668,6 +671,33 @@ static int fsl_sai_probe(struct platform_device *pdev) SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); } +#ifdef CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + /* Restore all registers */ + regcache_cache_only(sai->regmap, false); + regcache_sync(sai->regmap); + + return 0; +}; +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, @@ -680,6 +710,7 @@ static struct platform_driver fsl_sai_driver = { .name = "fsl-sai", .owner = THIS_MODULE, .of_match_table = fsl_sai_ids, + .pm = &fsl_sai_pm, }, }; module_platform_driver(fsl_sai_driver); -- 2.1.0.27.g96db324

4 5

[alsa-devel] [PATCH] ASoC: Intel: fix missing mutex
by Sudip Mukherjee 02 Nov '14

02 Nov '14

on error in block prepare, we were returning the error code while still holding the mutex. We are releasing the mutex in this patch before return. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/intel/sst-firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 35788ad..dca6d9a 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -1120,6 +1120,7 @@ int sst_block_alloc_scratch(struct sst_dsp *dsp) ret = block_list_prepare(dsp, &dsp->scratch_block_list); if (ret < 0) { dev_err(dsp->dev, "error: scratch block prepare failed\n"); + mutex_unlock(&dsp->mutex); return ret; } -- 1.8.1.2

2 1

Re: [alsa-devel] ASoC HDMI receiver driver
by Tim Harvey 31 Oct '14

31 Oct '14

On Thu, Oct 30, 2014 at 12:41 AM, Lars-Peter Clausen <lars(a)metafoo.de> wrote: > On 10/30/2014 04:34 AM, Tim Harvey wrote: >> Greetings, >> >> I'm working on an ASoC driver (an mfd driver for an HDMI receiver chip >> with an accompanying codec driver for the audio capture portion, and a >> cpu-dai driver for the specific SoC its hooked to). I can't seem to >> figure out what kind of hook I need in order to fill out the >> snd_pcm_substream fields specifying the rate/channels/sample-format of >> the receiving HDMI audio stream. In this case HDMI audio (and thus the >> codec) supports several rates but it doesn't make sense that the codec >> is told what rate to capture at as the rate is dictated by the current >> stream the receiver is being provided. I'm thinking there is some hook >> I'm missing that allows the codec to fill in the params dynamically >> when the capture device is opened. >> >> Any pointers to how I deal with this? > > In the startup() callback of the CODEC driver you can apply additional > constraints to the available parameter ranges. See > snd_pcm_hw_constraints_... in include/sound/pcm.h > > E.g. to constrain the rate to one fixed value you could do > snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, rate, rate); > > - Lars > Lars, Thanks for the tip! I found that if I add some validation into the hw_params function for the codec, that I can return an -EINVAL if the requested params don't match the incoming stream and that will cause the ALSA application to at least error out. However, when I saw your response I then also tried setting the constraint and found that an invalid sample-rate gets 'refined' to the available rate thus the hw_params sanity check passes, yet the user application apparently does not feedback that the rate has been changed. For example if I do the following on a capture stream which is actually 44.1kHz from the HDMI source with a constraint to set the rate to 44100 the audio is played back at 44100 and has all kinds of pops etc: gst-launch alsasrc device="my-hdmi-receiver-card" ! audio/x-raw-int,rate=48000 ! queue ! alsasink device="playback-card" Where-as the following plays back fine: gst-launch alsasrc device="my-hdmi-receiver-card" ! audio/x-raw-int,rate=44100 ! queue ! alsasink device="playback-card" I find that the value passed to startup in the substream is 48000 which gets refined to 44100 if the constraints were set, which ends up passing hw_params 44100 but the 'application' still has issues. I think its more appropriate to validate the hw_params instead of using the constraints. Does this make sense? Regards, Tim

2 3

[alsa-devel] [PATCH] ASoC: CS42L51_I2C depends on I2C
by Randy Dunlap 31 Oct '14

31 Oct '14

From: Randy Dunlap <rdunlap(a)infradead.org> Fix build errors when CONFIG_I2C is not enabled by making the driver depend on I2C. ../sound/soc/codecs/cs42l51-i2c.c:55:1: warning: data definition has no type or storage class [enabled by default] module_i2c_driver(cs42l51_i2c_driver); ^ ../sound/soc/codecs/cs42l51-i2c.c:55:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] ../sound/soc/codecs/cs42l51-i2c.c:55:1: warning: parameter names (without types) in function declaration [enabled by default] ../sound/soc/codecs/cs42l51-i2c.c:45:26: warning: 'cs42l51_i2c_driver' defined but not used [-Wunused-variable] static struct i2c_driver cs42l51_i2c_driver = { ^ Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20141031.orig/sound/soc/codecs/Kconfig +++ linux-next-20141031/sound/soc/codecs/Kconfig @@ -338,6 +338,7 @@ config SND_SOC_CS42L51 config SND_SOC_CS42L51_I2C tristate "Cirrus Logic CS42L51 CODEC (I2C)" + depends on I2C select SND_SOC_CS42L51 config SND_SOC_CS42L52

3 2

[alsa-devel] [PATCH 00/14] ASoC: AC'97 driver cleanups
by Lars-Peter Clausen 31 Oct '14

31 Oct '14

Hi, This series does a few AC'97 driver cleanups in preparation of the conversion of the drivers to regmap. Most of the patches are quite trivial. The last two patches remove the virtual registers from the wm9712 and wm9713 drivers, which is a requirement for regmap conversion. The last patch is the only one in this series that is a bit more complex. Currently the drivers have a open-coded implementation of shared controls which is replaced by the patch with the generic one from the ASoC core. Unfortunately this changes the controls that are exposed by the driver, instead of having a "Left HP Mixer ..." and a "Right HP Mixer ..." control for each mixer input there will only be a single "HP Mixer" control. This may or may not be considered breaking the userspace interface. If it is we need to take a different approach at removing the virtual register form the driver. - Lars Lars-Peter Clausen (14): ASoC: ad1980: Remove unused header ASoC: ad1980: Cleanup printk usage ASoC: ad1980: Use table based control setup ASoC: stac9766: Cleanup printk usage ASoC: stac9766: Use table based control setup ASoC: wm9705: Cleanup printk usage ASoC: wm9705: Use table based control setup ASoC: wm9712: Cleanup printk usage ASoC: wm9712: Use table based control setup ASoC: wm9713: Cleanup printk usage ASoC: wm9713: Use table based control setup ASoC: wm9713: Move driver state struct allocation to driver probe ASoC: wm9713: Use virtual control instead of virtual register ASoC: wm9712/wm9713: Use shared controls sound/soc/blackfin/bf5xx-ad1980.c | 2 - sound/soc/codecs/ad1980.c | 23 ++---- sound/soc/codecs/ad1980.h | 26 ------ sound/soc/codecs/stac9766.c | 9 +- sound/soc/codecs/wm9705.c | 14 ++-- sound/soc/codecs/wm9712.c | 136 +++++++----------------------- sound/soc/codecs/wm9713.c | 170 +++++++++++--------------------------- 7 files changed, 98 insertions(+), 282 deletions(-) delete mode 100644 sound/soc/codecs/ad1980.h -- 1.8.0

3 22

[alsa-devel] [PATCH 0/9] ASoC: Intel: more modulartion of init and add ACPI
by Vinod Koul 31 Oct '14

31 Oct '14

The first patch is a fix of mfld-pcm driver Rest of the series is devoted to ACPI support with 7 patch moving the probe routine and sperating into common and PCI and later moving PCI code to sst-pci file so that it can be shared with ACPI Lats patch adds support for ACPI with BYT driver support. This series is dependent upon the PM and compressed series sent earlier today Subhransu S. Prusty (9): ASoC: Intel: mfld-pcm: Fix to Store device context in sst_data ASoC: Intel: move the driver wq init to a routine ASoC: Intel: move the lock and wq initialization to routine ASoC: Intel: move the driver context allocation to routine ASoC: Intel: modularize driver probe and remove ASoC: Intel: more probe modularization for sst ASoC: Intel: move PCI probe to a seprate file ASoC: Intel: add shim save context and restore routines ASoC: Intel: Add ACPI driver for SST sound/soc/intel/Kconfig | 7 + sound/soc/intel/sst-mfld-platform-pcm.c | 1 + sound/soc/intel/sst/Makefile | 8 + sound/soc/intel/sst/sst.c | 391 ++++++++++++------------------- sound/soc/intel/sst/sst.h | 7 + sound/soc/intel/sst/sst_acpi.c | 387 ++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pci.c | 225 ++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 8 files changed, 793 insertions(+), 235 deletions(-) create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_pci.c

3 24

[alsa-devel] [PATCH] ASoC: Intel: Add jack detection for Broadwell
by Jie Yang 31 Oct '14

31 Oct '14

Add jack dectection and event reporting for Broadwell. It use combo jack on BDW platform, which including Mic Jack pin and Headphone jack pin. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 49 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 46 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 0e550f1..52cb764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -19,6 +19,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <sound/pcm_params.h> #include "sst-dsp.h" @@ -26,8 +27,26 @@ #include "../codecs/rt286.h" +static struct snd_soc_jack broadwell_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin broadwell_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broadwell_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), +}; + static const struct snd_soc_dapm_widget broadwell_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), SND_SOC_DAPM_MIC("DMIC1", NULL), @@ -42,7 +61,7 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"Speaker", NULL, "SPOL"}, /* HP jack connectors - unknown if we have jack deteck */ - {"Headphones", NULL, "HPO Pin"}, + {"Headphone Jack", NULL, "HPO Pin"}, /* other jacks */ {"MIC1", NULL, "Mic Jack"}, @@ -57,6 +76,27 @@ static const struct snd_soc_dapm_route broadwell_rt286_map[] = { {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, }; +static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + ret = snd_soc_jack_new(codec, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); + + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&broadwell_headset, + ARRAY_SIZE(broadwell_headset_pins), + broadwell_headset_pins); + if (ret) + return ret; + + rt286_mic_detect(codec, &broadwell_headset); + return 0; +} + + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -116,7 +156,7 @@ static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) } /* always connected - check HP for jack detect */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Line Jack"); @@ -196,6 +236,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .no_pcm = 1, .codec_name = "i2c-INT343A:00", .codec_dai_name = "rt286-aif1", + .init = broadwell_rt286_codec_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ignore_suspend = 1, @@ -213,6 +254,8 @@ static struct snd_soc_card broadwell_rt286 = { .owner = THIS_MODULE, .dai_link = broadwell_rt286_dais, .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .controls = broadwell_controls, + .num_controls = ARRAY_SIZE(broadwell_controls), .dapm_widgets = broadwell_widgets, .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), .dapm_routes = broadwell_rt286_map, -- 1.9.1

2 1

[alsa-devel] [PATCH TINYCOMPRESS] makefile.linux: allow env to override CROSS_COMPILE and CC
by Qais Yousef 31 Oct '14

31 Oct '14

allow env variables to override them without having to modify the makefile. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> --- makefile.linux | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/makefile.linux b/makefile.linux index 81782d0af461..c4dab6d3b7d5 100644 --- a/makefile.linux +++ b/makefile.linux @@ -9,8 +9,8 @@ LIBOBJ = ${LIBSRC:.c=.o} PLAYOBJ = ${PLAYSRC:.c=.o} RECOBJ = ${RECSRC:.c=.o} -CC = gcc -CROSS_COMPILE = +CROSS_COMPILE ?= +CC ?= ${CROSS_COMPILE}gcc CFLAGS += -std=c99 -Wall -Wextra -Wunused -DVERSION=\"${VER}\" -I./include LDFLAGS += -L. -ltinycompress @@ -20,19 +20,19 @@ all: libtinycompress cplay crec $(LIB): ${LIBOBJ} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} + @${CC} ${CFLAGS} -shared -Wl,-soname,$@.so -o $@.so ${LIBOBJ} ${LIBLDFLAGS} $(PLAYBIN): ${PLAYOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${PLAYOBJ} ${LDFLAGS} $(RECBIN): ${RECOBJ} ${LIB} @echo " LD "$@ - @${CROSS_COMPILE}${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} + @${CC} ${CFLAGS} -o $@ ${RECOBJ} ${LDFLAGS} %.o: %.c @echo " CC "$< - @${CROSS_COMPILE}${CC} ${CFLAGS} -c -fPIC -o $@ $< + @${CC} ${CFLAGS} -c -fPIC -o $@ $< clean: @rm -rf ${BIN} ${PLAYOBJ} ${RECOBJ} ${LIB}.so ${LIBOBJ} -- 2.1.0

1 0

[alsa-devel] [PATCH v2 0/3] ASoC: Intel: add runtime power management support
by Vinod Koul 31 Oct '14

31 Oct '14

This series fixes the device open pm handling and adds the runtime pm support. Also we fix lock usage while changing the driver state Changes in v2: - fix style in comments - fix error prints and remove noisy debug Mythri P K (1): Audio: SST: use lock when changing SST state. Subhransu S. Prusty (1): ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream Vinod Koul (1): ASoC: Intel: sst: add runtime power management handling sound/soc/intel/sst/sst.c | 65 +++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_drv_interface.c | 12 ++---- sound/soc/intel/sst/sst_ipc.c | 2 +- 3 files changed, 70 insertions(+), 9 deletions(-)

2 8

[alsa-devel] [PATCH 0/3] ASoC: Intel: add runtime power management support
by Vinod Koul 31 Oct '14

31 Oct '14

This series fixes the device opne pm handlinga dn adds the runtime pm support. Also we fix lock usage while changing teh driver state Mythri P K (1): Audio: SST: use lock when changing SST state. Subhransu S. Prusty (1): ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream Vinod Koul (1): ASoC: Intel: sst: add runtime power management handling sound/soc/intel/sst/sst.c | 71 ++++++++++++++++++++++++++++++- sound/soc/intel/sst/sst_drv_interface.c | 12 ++---- sound/soc/intel/sst/sst_ipc.c | 2 +- 3 files changed, 75 insertions(+), 10 deletions(-)

2 6

[alsa-devel] [PATCH] ASoC: TLV320AIC23: make codecs selectable in Kconfig
by Max Filippov 31 Oct '14

31 Oct '14

Now that manual selection of drivers for audio subsystem components is preferred AIC23 codec must be selectable in Kconfig to make it possible. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8838838e..7dad805 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -552,11 +552,11 @@ config SND_SOC_TLV320AIC23 tristate config SND_SOC_TLV320AIC23_I2C - tristate + tristate "Texas Instruments TLV320AIC23 audio CODEC - I2C" select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC23_SPI - tristate + tristate "Texas Instruments TLV320AIC23 audio CODEC - SPI" select SND_SOC_TLV320AIC23 config SND_SOC_TLV320AIC26 -- 1.8.1.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix block is enabled multiple times issue
by Jie Yang 31 Oct '14

31 Oct '14

During FW parsing and loading, block_list_prepare() may be called for each raw data block copying and this may made the hsw_block_enable() called mutiple times, which increase block->users many times. The result of this is hsw_block_disable() can't power gated the related block when trying to free the blocks during suspend, and the power gating status also confused. Here check the block user status, only calling enable() for those blocks who has no user yet. Remember that this works correctlly on current case, where there are enough SRAM memory so different module won't share a memory block. For further usage, we may need restructure the struct sst_mem_block to save the module list who is using it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 35788ad..c451398 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -149,7 +149,7 @@ static int block_list_prepare(struct sst_dsp *dsp, /* enable each block so that's it'e ready for data */ list_for_each_entry(block, block_list, module_list) { - if (block->ops && block->ops->enable) { + if (block->ops && block->ops->enable && !block->users) { ret = block->ops->enable(block); if (ret < 0) { dev_err(dsp->dev, -- 1.9.1

3 3

Re: [alsa-devel] Restore hidden register for default value
by Kailang 31 Oct '14

31 Oct '14

Hi Takashi, It's just the default value refill in initial state. It's only for ALC668(ALC3661). The patch was tested pass with issue machine in ODM site. Many Thanks. BR, Kailang > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, October 22, 2014 4:00 PM > To: Kailang > Subject: Re: Restore hidden register for default value > > At Wed, 22 Oct 2014 07:56:43 +0000, > Kailang wrote: > > > > Hi Takashi, > > > > We want to post ALC668 for hidden register issue. > > > > Could I post like below array for codec initial state? > > > > WORD coefIndexDefault[][2] = {{0x1, > 0xbebe},{0x2,0xaaaa},{0x3,0x0},{0x4,0x180},{0x6,0x0}, > > > {0x7,0xf80},{0x8,0x31},{0xa,0x60},{0xb,0x0},{0xc,0x7cf7},{0xd, > 0x1080},{0xe,0x7f7f}, > > > {0xf,0xcccc},{0x10,0xddcc},{0x11,0x1},{0x13,0x0},{0x14,0x2aa0} > ,{0x17,0xA940}, > > > {0x19,0},{0x1A,0},{0x1B,0},{0x1c,0x0},{0x1d,0x0},{0x1e,0x7418} > ,{0x1f,0x804}, > > > {0x20,0x4200},{0x21,0x468},{0x22,0x8ccc},{0x23,0x250},{0x24,0x7418}, > > > > > {0x27,0x0000},{0x28,0x8ccc},{0x2a,0xff00},{0x2b,0x8000},{0xa7,0xff00}} > > ; Because codec sometimes has wrong value for hidden register after > > resume back or reboot. > > It needed to refill the default value with upper array in > initial state. > > But the hidden registers was more for 59 records. > > Could I post it with no remark? Because it just the default > value. Thanks. > > Some comments would be nice, of course, but not mandatory. > > One concern is exactly which codecs should this be applied. > There are a bunch of codecs compatible with alc668. It's > only for ALC668, and not for others (ALC668, 670, whatever). > > > Takashi > > ------Please consider the environment before printing this e-mail. >

1 0

[alsa-devel] [PATCH] usb-audio: Duplex support for Digidesign Mbox 1 and clock source selector
by Damien Zammit 31 Oct '14

31 Oct '14

Hi, Please review this patch and my previous one. This patch provides duplex support for the Digidesign Mbox 1 sound card. This also depends on my previous patch for enabling the clock source selector because the mixer is enabled in this patch. I reduced the supported sampling rates to just 48kHz because the device works better in this mode with the SPDIF external sync. Damien

1 0

[alsa-devel] [PATCH] usb-audio: Add mixer control for Digidesign Mbox 1 clock source
by Damien Zammit 31 Oct '14

31 Oct '14

This patch provides a mixer control for selecting the clock source of the Digidesign Mbox 1 to either internal clock or S/PDIF external. Trial and error and bus snooping were the only way to get this information, but it works on the hardware.

2 5

[alsa-devel] [PATCH v2 2/2] [media] sound: Update au0828 quirks table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

The au0828 quirks table is currently not in sync with the au0828 media driver. Syncronize it and put them on the same order as found at au0828 driver, as all the au0828 devices with analog TV need the same quirks. Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8f3e2bf100eb..83bddbdb90e9 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2827,14 +2827,22 @@ YAMAHA_DEVICE(0x7010, "UB99"), } AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), -AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7201, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7211, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7281, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x05e1, 0x0480, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x8200, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x7260, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7213, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

3 5

[alsa-devel] [PATCH] ASoC: Intel: fix build with runtime PM disabled.
by Liam Girdwood 30 Oct '14

30 Oct '14

Fix the following errors: All error/warnings: >> sound/soc/intel/sst-haswell-pcm.c:1168:13: error: 'hsw_pcm_prepare' undeclared here (not in a function) .prepare = hsw_pcm_prepare, ^ >> sound/soc/intel/sst-haswell-pcm.c:1169:14: error: 'hsw_pcm_complete' undeclared here (not in a function) .complete = hsw_pcm_complete, ^ Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 14 ++++++-------- 2 files changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 0ea7c3d..ffd5728 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1711,7 +1711,7 @@ void sst_hsw_runtime_module_free(struct sst_module_runtime *runtime) sst_module_runtime_free(runtime); } -#ifdef CONFIG_PM_RUNTIME +#ifdef CONFIG_PM static int sst_hsw_dx_state_dump(struct sst_hsw *hsw) { struct sst_dsp *sst = hsw->dsp; diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 4489a35..cd54dd9 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -1058,6 +1058,12 @@ static int hsw_pcm_runtime_resume(struct device *dev) return ret; } +#else +#define hsw_pcm_runtime_idle NULL +#define hsw_pcm_runtime_suspend NULL +#define hsw_pcm_runtime_resume NULL +#endif + static void hsw_pcm_complete(struct device *dev) { struct hsw_priv_data *pdata = dev_get_drvdata(dev); @@ -1153,14 +1159,6 @@ static int hsw_pcm_prepare(struct device *dev) return 0; } -#else -#define hsw_pcm_runtime_idle NULL -#define hsw_pcm_runtime_suspend NULL -#define hsw_pcm_runtime_resume NULL -#define hsw_pcm_runtime_complete NULL -#define hsw_pcm_runtime_prepare NULL -#endif - static const struct dev_pm_ops hsw_pcm_pm = { .runtime_idle = hsw_pcm_runtime_idle, .runtime_suspend = hsw_pcm_runtime_suspend, -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: Intel; Add debug output when boot fails.
by Liam Girdwood 30 Oct '14

30 Oct '14

Add the debug output from IPCD and IPCX when booting fails. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b37d3ee..0ea7c3d 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1927,7 +1927,9 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { - dev_err(hsw->dev, "error: audio DSP boot timeout\n"); + dev_err(hsw->dev, "error: audio DSP boot timeout IPCD 0x%x IPCX 0x%x\n", + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCD), + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX)); return -EIO; } @@ -2038,7 +2040,9 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) msecs_to_jiffies(IPC_BOOT_MSECS)); if (ret == 0) { ret = -EIO; - dev_err(hsw->dev, "error: ADSP boot timeout\n"); + dev_err(hsw->dev, "error: audio DSP boot timeout IPCD 0x%x IPCX 0x%x\n", + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCD), + sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX)); goto boot_err; } -- 2.1.0

2 1

[alsa-devel] opengl 'leaking' sound to input.
by dE 30 Oct '14

30 Oct '14

I'm trying to record audio from the analog input device of my card, but whenever there seems to be some graphics intensive activity like glxgears, OpenGL assisted window manager, audacity graphs etc... a noise appears in the recording which sounds like some kind of sparking disturbance. I've recorded audio many times on this box before, but this started to happen suddenly (maybe after an ugprade, but old live CDs with proprietary nvidia drivers installed or newer nouveau drivers seem to have the same issue). Just to confirm, is this a hardware issue?

2 2

[alsa-devel] [ALSA 0/8] Yet more documentation updates
by Takashi Iwai 30 Oct '14

30 Oct '14

Here are more updates of patches to complete kerneldoc comments. At this time, HD-audio generic parser, control PCM and core stuff are covered. Also, ASoC files are included in alsa-driver-api documents now. [PATCH 1/8] ALSA: hda - Add kerneldoc comments to hda_generic.c [PATCH 2/8] ALSA: doc: Add ASoC codes into API documentation [PATCH 3/8] ALSA: control: Add missing kerneldoc comments to exported [PATCH 4/8] ALSA: pcm: Use static inline for [PATCH 5/8] ALSA: pcm: More kerneldoc updates [PATCH 6/8] ALSA: pcm: Convert params_*() with static inline [PATCH 7/8] ALSA: doc: More kerneldoc comments on core components [PATCH 8/8] ALSA: doc: Fix missing "I" for kerneldoc inclusion Takashi

1 8

[alsa-devel] [PATCH] ASoC: Intel: Work around to fix HW D3 potential crash issue
by Jie Yang 30 Oct '14

30 Oct '14

When using clock gatings to save power, there are some known issues: 1. core clock gating (DCLCGE) must be disabled during D0 and D3 entry and updating SRAM banks (VDRTCTL0). 2. DSP trunk clock gating (DTCGE) can cause FW crashes, disable it in D0. To align with the new W/A flow from FW team, we must set VDRTCTL0.D3PGD to 1 (D3 power gating disabled) at first startup and keep it all the time. ADSP will be in D0 on first boot by BIOS part of WA. Required delays must be preserved (waiting for HW to stabilize, after enabling CCG, changing SRAM PG, D3PG). D3->D0: 1. Disable core clock gating (VDRTCTL2.DCLCGE = 0) 2. Enable other CG apart from DTCG and DCLCG (VDRTCTL2. DCLCGE and DTCGE = 0) 3. Disable D3PG (VDRTCTL0.D3PGD = 1) 4. Power up necessary SRAM and wait at least for 18 clock cycles for every bank you have powered up 5. Set D0 state(PMCS.PS = 0), wait for HW 6. Restore MCLK (clkctl.smos, disabled in D3 entry point 4) 7. Stall and reset core, set CSR 8. Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us 9. Unreset core 10.Load FW, configure PLL and other necessary things 11.Unstall core Changing SRAM PG during D0: 1. Disable core clock gating (VDRTCTL2.DCLCGE = 0) 2. Set PG mask 3. Wait at least for 18 clock cycles for every bank you have powered up 4. Enable core clock gating, delay 50 us D0->D3: 1. Disable core clock gating (DCLCGE = 0) 2. Stall and reset core 3. Power down entire SRAM and wait at least for 18 clock cycles for every bank (Enable SRAM PG (ISRAMPGE = 0x3FF, DSRAMPGE = 0xFFFFF, D3SRAMPGD = 0), remember about preserving VDRTCTL0.D3PGD = 1) 4. Shutdown PLL, disable MCLK(clkctl.smos = 0), Enable DTCG to save power 5. Set D3 state(PMCS.PS = 3), delay 50 us 6. Enable core clock gating, delay 50 us Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-dsp.h | 14 ++++-- sound/soc/intel/sst-haswell-dsp.c | 102 ++++++++++++++++++++++++++++++++------ 2 files changed, 98 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index 2753b85..f291e32 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -159,12 +159,18 @@ #define SST_VDRTCTL3 0xaC /* VDRTCTL0 */ -#define SST_VDRTCL0_APLLSE_MASK 1 -#define SST_VDRTCL0_DSRAMPGE_SHIFT 16 -#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) -#define SST_VDRTCL0_ISRAMPGE_SHIFT 6 +#define SST_VDRTCL0_D3PGD (1 << 0) +#define SST_VDRTCL0_D3SRAMPGD (1 << 1) +#define SST_VDRTCL0_DSRAMPGE_SHIFT 12 +#define SST_VDRTCL0_DSRAMPGE_MASK (0xfffff << SST_VDRTCL0_DSRAMPGE_SHIFT) +#define SST_VDRTCL0_ISRAMPGE_SHIFT 2 #define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) +/* VDRTCTL2 */ +#define SST_VDRTCL2_DCLCGE (1 << 1) +#define SST_VDRTCL2_DTCGE (1 << 10) +#define SST_VDRTCL2_APLLSE_MASK (1 << 31) + /* PMCS */ #define SST_PMCS 0x84 #define SST_PMCS_PS_MASK 0x3 diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index 86aea34..57039b0 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -250,17 +250,42 @@ static irqreturn_t hsw_irq(int irq, void *context) static void hsw_set_dsp_D3(struct sst_dsp *sst) { u32 val; + u32 reg; + + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* switch off audio PLL, DRAM & IRAM blocks */ + /* enable power gating and switch off DRAM & IRAM blocks */ val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - val |= SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | + val |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); - /* Set D3 state */ + /* switch off audio PLL */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_APLLSE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* disable MCLK(clkctl.smos = 0) */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK, 0); + + /* Set D3 state, delay 50 us */ val = readl(sst->addr.pci_cfg + SST_PMCS); val |= SST_PMCS_PS_MASK; writel(val, sst->addr.pci_cfg + SST_PMCS); + udelay(50); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + } static void hsw_reset(struct sst_dsp *sst) @@ -283,6 +308,16 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) int tries = 10; u32 reg; + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* Set D0 state */ reg = readl(sst->addr.pci_cfg + SST_PMCS); reg &= ~SST_PMCS_PS_MASK; @@ -300,14 +335,6 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) return -ENODEV; finish: - hsw_reset(sst); - - /* switch on audio PLL, DRAM & IRAM blocks */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg &= ~(SST_VDRTCL0_APLLSE_MASK | SST_VDRTCL0_DSRAMPGE_MASK | - SST_VDRTCL0_ISRAMPGE_MASK); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); @@ -322,6 +349,28 @@ finish: SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + /* Stall and reset core, set CSR */ + hsw_reset(sst); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + + /* switch on audio PLL */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg &= ~SST_VDRTCL2_APLLSE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, SST_CSR2_SDFD_SSP1); @@ -343,9 +392,6 @@ finish: sst_dsp_shim_write(sst, 0x80, 0x6); sst_dsp_shim_write(sst, 0xe0, 0x300a); - /* disable all clock gating */ - writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); - return 0; } @@ -497,6 +543,11 @@ static int hsw_block_enable(struct sst_mem_block *block) dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n", block->type, block->index, block->offset); + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val &= ~SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0); @@ -504,6 +555,13 @@ static int hsw_block_enable(struct sst_mem_block *block) /* wait 18 DSP clock ticks */ udelay(10); + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/ sst_mem_block_dummy_read(block); return 0; @@ -521,10 +579,26 @@ static int hsw_block_disable(struct sst_mem_block *block) dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n", block->type, block->index, block->offset); + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val &= ~SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); bit = hsw_block_get_bit(block); writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* wait 18 DSP clock ticks */ + udelay(10); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_DCLCGE; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + + udelay(50); + return 0; } -- 1.9.1

1 0

[alsa-devel] Build error on Fedora 21
by Mark van Rossum 30 Oct '14

30 Oct '14

Hi alsa_utils_alsa-compile.sh fails with: make[1]: Entering directory '/usr/src/kernels/3.17.0-301.fc21.x86_64' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.c: In function ‘snd_info_version_read’: /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.c:1065:22: error: macro "__DATE__" might prevent reproducible builds [-Werror=date-time] "Compiled on " __DATE__ " for kernel %s" ^ cc1: some warnings being treated as errors scripts/Makefile.build:257: recipe for target '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o' failed make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore/info.o] Error 1 scripts/Makefile.build:404: recipe for target '/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore' failed make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d/acore] Error 2 Makefile:1373: recipe for target '_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d' failed make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d] Error 2 make[1]: Leaving directory '/usr/src/kernels/3.17.0-301.fc21.x86_64' Makefile:167: recipe for target 'compile' failed make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.25.3584.gb044d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. -- The University of Edinburgh is a charitable body, registered in Scotland, with registration number SC005336.

2 1

[alsa-devel] [PATCH 1/2] [media] sound: simplify au0828 quirk table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

From: Mauro Carvalho Chehab <m.chehab(a)samsung.com> Add a macro to simplify au0828 quirk table. That makes easier to check it against the USB IDs at drivers/media/usb/au0828-card.c Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <m.chehab(a)samsung.com> Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752a420c..5ae1d02d17a3 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2804,133 +2804,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, -/* Hauppauge HVR-950Q and HVR-850 */ -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-850", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7280), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x0fd9, 0x0008), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, +/* + * Auvitek au0828 devices with audio interface. + * This should be kept in sync with drivers/media/usb/au0828-card.c + * Please notice that some drivers are DVB only, and don't need to be + * here. That's the case, for example, of DVICO_FUSIONHDTV7. + */ + +#define AU0828_DEVICE(vid, pid, vname, pname) { \ + USB_DEVICE_VENDOR_SPEC(vid, pid), \ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ + USB_DEVICE_ID_MATCH_INT_CLASS | \ + USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ + .bInterfaceClass = USB_CLASS_AUDIO, \ + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, \ + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { \ + .vendor_name = vname, \ + .product_name = pname, \ + .ifnum = QUIRK_ANY_INTERFACE, \ + .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + } \ +} + +AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), +AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

3 4

[alsa-devel] [PATCH v2 1/2] [media] sound: simplify au0828 quirk table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

Add a macro to simplify au0828 quirk table. That makes easier to check it against the USB IDs at drivers/media/usb/au0828/au0828-cards.c. Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/drivers/media/usb/au0828/au0828-cards.c b/drivers/media/usb/au0828/au0828-cards.c index 9eb77ac2153b..da87f1cc31a9 100644 --- a/drivers/media/usb/au0828/au0828-cards.c +++ b/drivers/media/usb/au0828/au0828-cards.c @@ -36,6 +36,11 @@ static void hvr950q_cs5340_audio(void *priv, int enable) au0828_clear(dev, REG_000, 0x10); } +/* + * WARNING: There's a quirks table at sound/usb/quirks-table.h + * that should also be updated every time a new device with V4L2 support + * is added here. + */ struct au0828_board au0828_boards[] = { [AU0828_BOARD_UNKNOWN] = { .name = "Unknown board", diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752a420c..8f3e2bf100eb 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2804,133 +2804,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, -/* Hauppauge HVR-950Q and HVR-850 */ -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7210), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7217), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721b), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721e), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x721f), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7240), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-850", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x2040, 0x7280), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, -{ - USB_DEVICE_VENDOR_SPEC(0x0fd9, 0x0008), - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .vendor_name = "Hauppauge", - .product_name = "HVR-950Q", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_ALIGN_TRANSFER, - } -}, +/* + * Auvitek au0828 devices with audio interface. + * This should be kept in sync with drivers/media/usb/au0828/au0828-cards.c + * Please notice that some drivers are DVB only, and don't need to be + * here. That's the case, for example, of DVICO_FUSIONHDTV7. + */ + +#define AU0828_DEVICE(vid, pid, vname, pname) { \ + USB_DEVICE_VENDOR_SPEC(vid, pid), \ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ + USB_DEVICE_ID_MATCH_INT_CLASS | \ + USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ + .bInterfaceClass = USB_CLASS_AUDIO, \ + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, \ + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { \ + .vendor_name = vname, \ + .product_name = pname, \ + .ifnum = QUIRK_ANY_INTERFACE, \ + .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + } \ +} + +AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), +AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

1 0

[alsa-devel] [PATCH 2/2] [media] sound: Update au0828 quirks table
by Mauro Carvalho Chehab 30 Oct '14

30 Oct '14

From: Mauro Carvalho Chehab <m.chehab(a)samsung.com> The au0828 quirks table is currently not in sync with the au0828 media driver. Syncronize it and put them on the same order as found at au0828 driver, as all the au0828 devices with analog TV need the same quirks. Cc: stable(a)vger.kernel.org Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5ae1d02d17a3..8a6b366f2925 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2827,14 +2827,22 @@ YAMAHA_DEVICE(0x7010, "UB99"), } AU0828_DEVICE(0x2040, 0x7200, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7210, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x7217, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721b, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721e, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x2040, 0x721f, "Hauppauge", "HVR-950Q"), -AU0828_DEVICE(0x2040, 0x7240, "Hauppauge", "HVR-850"), AU0828_DEVICE(0x2040, 0x7280, "Hauppauge", "HVR-950Q"), AU0828_DEVICE(0x0fd9, 0x0008, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7201, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7211, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x2040, 0x7281, "Hauppauge", "HVR-950Q-MXL"), +AU0828_DEVICE(0x05e1, 0x0480, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x8200, "Hauppauge", "Woodbury"), +AU0828_DEVICE(0x2040, 0x7260, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7213, "Hauppauge", "HVR-950Q"), +AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), /* Digidesign Mbox */ { -- 1.9.3

1 0

[alsa-devel] Selecting correct input device port.
by dE 30 Oct '14

30 Oct '14

As I learn't in this mailng list, a device can typically handle only one stream at a time unless it supports hardware mixing which's exposed via subdevices in ALSA What does subdevice mean for input device? Do they mark physical ports for input? If not how do I record from a specific input port? Is it automatically selected by the card? And what's 'input source' in ALSA mixer?

3 6

Re: [alsa-devel] Update Initial AMP for EAPD control
by Takashi Iwai 30 Oct '14

30 Oct '14

At Thu, 30 Oct 2014 07:38:08 +0000, Kailang wrote: > > Hi Takashi, > > Attach patch is update initial AMP for EAPD control. > > Change log: > - alc889_coef_init() is replaced with alc_update_coef_idx() with a > correct COEF value. > - for ALC262, ALC887 and ALC900, the EAPD setup via the hidden > register is removed because this rather conflicts with the EAPD verb > setup. > - For ALC888-VC, also the hidden register access is removed in > alc888_coef_init(). > - Remove the dead #if 0 code for ALC267/ALC268. > > This update will avoid wrong behavior for some codec. > This patch will fix double setup for EAPD. > It just need to turn on by one site for verb command or hidden register controlled. I found that there is one caller of alc889_coef_init(), so I modified it accordingly, too. Also fixed typos in the changelog, and edited a bit, then applied to git tree. thanks, Takashi > > Many Thanks. > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0001-update-init-amp-for-eapd.patch <application/octet-stream (base64)>] >

1 0