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[alsa-devel] [PATCH] ASoC: fsi: reserve prefetch period on DMA transferring
by Kuninori Morimoto 26 Aug '13

26 Aug '13

Current FSI is supporting DMAEngine transfer, but, it needs to use work queue. Therefore, DMA transfer settings might be late if there is heavy task. This patch reserves next period beforehand on DMA transfer function. Android sound will be breaking up without this patch. Tested-by: Tomohito Esaki <etom(a)igel.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 51 +++++++++++++++++++++++++++++++++------------------ 1 file changed, 33 insertions(+), 18 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4fc2a2a..c608c22 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -235,6 +235,8 @@ struct fsi_stream { struct sh_dmae_slave slave; /* see fsi_handler_init() */ struct work_struct work; dma_addr_t dma; + int loop_cnt; + int additional_pos; }; struct fsi_clk { @@ -1383,6 +1385,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */ + io->additional_pos = 0; io->dma = dma_map_single(dai->dev, runtime->dma_area, snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; @@ -1399,11 +1403,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static dma_addr_t fsi_dma_get_area(struct fsi_stream *io) +static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional) { struct snd_pcm_runtime *runtime = io->substream->runtime; + int period = io->period_pos + additional; - return io->dma + samples_to_bytes(runtime, io->buff_sample_pos); + if (period >= runtime->periods) + period = 0; + + return io->dma + samples_to_bytes(runtime, period * io->period_samples); } static void fsi_dma_complete(void *data) @@ -1415,7 +1423,7 @@ static void fsi_dma_complete(void *data) enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io), + dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0), samples_to_bytes(runtime, io->period_samples), dir); io->buff_sample_pos += io->period_samples; @@ -1441,7 +1449,7 @@ static void fsi_dma_do_work(struct work_struct *work) struct snd_pcm_runtime *runtime; enum dma_data_direction dir; int is_play = fsi_stream_is_play(fsi, io); - int len; + int len, i; dma_addr_t buf; if (!fsi_stream_is_working(fsi, io)) @@ -1451,26 +1459,33 @@ static void fsi_dma_do_work(struct work_struct *work) runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); - buf = fsi_dma_get_area(io); - dma_sync_single_for_device(dai->dev, buf, len, dir); + for (i = 0; i < io->loop_cnt; i++) { + buf = fsi_dma_get_area(io, io->additional_pos); - desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } + dma_sync_single_for_device(dai->dev, buf, len, dir); - desc->callback = fsi_dma_complete; - desc->callback_param = io; + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } - if (dmaengine_submit(desc) < 0) { - dev_err(dai->dev, "tx_submit() fail\n"); - return; + desc->callback = fsi_dma_complete; + desc->callback_param = io; + + if (dmaengine_submit(desc) < 0) { + dev_err(dai->dev, "tx_submit() fail\n"); + return; + } + + dma_async_issue_pending(io->chan); + + io->additional_pos = 1; } - dma_async_issue_pending(io->chan); + io->loop_cnt = 1; /* * FIXME -- 1.7.9.5

2 1

[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 26 Aug '13

26 Aug '13

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

2 3

[alsa-devel] [RFC PATCH 1/3] ALSA: hda - add flags and routines to get devices selection info for DP1.2 MST
by mengdong.lin＠intel.com 26 Aug '13

26 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds flags and routines to get device list & selection info on a pin. To support Display Port 1.2 multi-stream transport (MST) over single DP port, a pin can support multiple devices. Please refer to HD-A spec Document Change Notificaton HDA040-A. A display audio codec can set flag "dp_mst" in its patch, indicating its pins can support MST. But at runtime, a pin may not be multi-streaming capable and report the device list is empty, depending on Gfx driver configuration. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fdbb09a..fb58291 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +666,63 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEVLIST_LEN; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + union hda_devlist_entry *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + devices = 0; + + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices].value = parm & 0x7; + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e0..736b55a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -795,6 +799,19 @@ struct hda_pcm { bool own_chmap; /* codec driver provides own channel maps */ }; +/* display pin's device list entry */ +union hda_devlist_entry { + unsigned int value; + struct { + unsigned int present_detect:1; + unsigned int eld_valid:1; + unsigned int inactive:1; + } bits; +}; + +#define AC_MAX_DEVLIST_LEN 64 +#define AC_DEVLIST_LEN 0x3f + /* codec information */ struct hda_codec { struct hda_bus *bus; @@ -885,6 +902,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -972,6 +990,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + union hda_devlist_entry *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); -- 1.8.1.2

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[alsa-devel] [PATCH v2 4/4] ALSA: hda - add device entry and inactive flag to unsolicited response
by mengdong.lin＠intel.com 25 Aug '13

25 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch adds two fields to unsolicited response, according to spec HDA040-A: - Device Entry (bit 20:15) - Inactive (bit 2) and show the info in debug message. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 316076e..c578f5e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -255,6 +255,11 @@ enum { #define AC_UNSOL_RES_TAG_SHIFT 26 #define AC_UNSOL_RES_SUBTAG (0x1f<<21) #define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ #define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ #define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ #define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 87ca984..895a0d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) int pin_nid; int pin_idx; struct hda_jack_tbl *jack; + int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) @@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, - "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, + "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v2 3/4] ALSA: hda - Haswell codec exposes device list/select info on pins
by mengdong.lin＠intel.com 25 Aug '13

25 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is only to allow codec proc file to expose devices list/select info for Haswell codec pins. Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list is empty, meaning no pin is multi-streaming capaple. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..87ca984 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,8 +1989,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v2 2/4] ALSA: hda - add device list & select info of display pins to codec proc file
by mengdong.lin＠intel.com 25 Aug '13

25 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If a display codec supports multi-stream transport on the pins, the pin's device list length and device entries will be exposed to codec proc file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 9760f00..1806250 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +582,38 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + struct hda_devlist_entry dev_list[AC_MAX_DEVLIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEVLIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %ld, ELDV = %ld, IA = %ld\n", i, + dev_list[i].present_detect, dev_list[i].eld_valid, + dev_list[i].inactive); + } + + return 0; + +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -751,6 +783,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); -- 1.8.1.2

1 0

[alsa-devel] [PATCH v5 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 24 Aug '13

24 Aug '13

Changelog: v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 76 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1336 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1817 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

6 36

[alsa-devel] [PATCH 1/2] ASoC: fsl_spdif: Remove unnecessary dev_set_drvdata()
by Fabio Estevam 23 Aug '13

23 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 42a4382..bf58086 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1184,7 +1184,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error_dev; + return ret; } ret = imx_pcm_dma_init(pdev); @@ -1197,8 +1197,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) error_component: snd_soc_unregister_component(&pdev->dev); -error_dev: - dev_set_drvdata(&pdev->dev, NULL); return ret; } -- 1.8.1.2

1 1

[alsa-devel] [PATCH] ASoC: simple-card: Provide owner and MODULE_ALIAS()
by Fabio Estevam 23 Aug '13

23 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Add .owner field and also MODULE_ALIAS(), so that auto module loading can work. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6cf8355..8c49147 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .owner = THIS_MODULE, }, .probe = asoc_simple_card_probe, .remove = asoc_simple_card_remove, @@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = { module_platform_driver(asoc_simple_card); +MODULE_ALIAS("platform:asoc-simple-card"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("ASoC Simple Sound Card"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 1.8.1.2

1 0

[alsa-devel] [PATCH V3] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 23 Aug '13

23 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- I removed RT5640_[LR]_MUTE bits control in rt5640_hp_event. And moved the required delay for hp depop to rt5640_hp_post_event. This delay time is for waitting for the codec finish the depop process. You will not haer anything if you play a very short sound without this delay. So I put it in post_event. And I think it is fine if hp is unmute during playback without this delay. I use rt5640->hp_mute to record the "HP Amp"'s status. However, I think it is better to use a dapm API for that purpose. But I don't know if there is a suitable API for it. Can anyone give me some advice? Thanks in advance. --- sound/soc/codecs/rt5640.c | 217 ++++++++++++++++++++++++++++++++++------------ sound/soc/codecs/rt5640.h | 12 +++ 2 files changed, 175 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..c26a8f8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_R_MUTE_SFT, 1, 1); + /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(10000, 15000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2 , + RT5640_PWR_FV1 | RT5640_PWR_FV2); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK, + RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN); + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + + regmap_write(rt5640->regmap, RT5640_PR_BASE + + RT5640_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK, + RT5640_HP_CP_PD | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + rt5640->hp_mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640->hp_mute = 1; + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_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: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->hp_mute) + usleep_range(80000, 85000); + + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), + SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1381,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, @@ -1396,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, - {"HPOL", NULL, "HP L Amp"}, - {"HPOR", NULL, "HP R Amp"}, + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, {"MONOP", NULL, "Mono MIX"}, @@ -1792,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(10000, 15000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); @@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, rt5640_dai, ARRAY_SIZE(rt5640_dai)); if (ret < 0) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..5e8df25a 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,7 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool hp_mute; }; #endif -- 1.8.1.1.439.g50a6b54

3 4

Re: [alsa-devel] usb-audio: Reloop Play support (TI TUSB3200AC)
by Daniel Mack 23 Aug '13

23 Aug '13

[Please do not drop the mailing list address] On 19.10.2012 15:45, Didier 'Ptitjes' Villevalois wrote: > On ven., 2012-10-19 at 15:20 +0200, Daniel Mack wrote: >> On 19.10.2012 15:09, Didier 'Ptitjes' Villevalois wrote: >>> Hello, >>> >>> I've bought a Reloop Play usb soundcard and I'd like to make it work >>> with ALSA. >>> >>> It is a 4-channels USB soundcard. It is currently correctly recognized >>> by ALSA as a 4-channels soundcard and I can correctly open the channels >>> and feed them with audio. However, no audio is output by the box. >> >> You're most probably affected by a regression in the driver that was >> fixes in 3.6.0 and 3.5.5 (and all later kernels). Could you please try >> one of these and tell us if that works? > > Sure, I'll do that now (I'm currently using 3.5.4 + mbp8,3 patches). > I guess 3.5.7 is OK ? I'm building it now and will reboot on that. Any kernel >= 3.6.0 and 3.5.5 should be ok. >>> It seems to contain a Texas Instrument TUSB3200AC chipset (same chipset >>> as the Maya44 USB and Hercules DJ console). I guess the problem is a >>> firmware problem but I don't know what to do about it. >>> >>> I've got a MacOS X driver that works correctly, if that of any help. (I >>> suppose the Windows driver works too but I don't have any Windows at >>> hand.) >> >> Did you install any vendor driver on OS X or does it just work with the >> one that ships with the OS? > > Yep, I had to install the Vendor-provided driver. What was the effect without that special driver? Could you see the device listed in the AudioMIDISetup? Could you stream audio? IOW, is the effect you see there comparable to how your Linux box behaves? You could try and remove that driver again and see what happens. > But it seems the > built-in USB soundcard support on MacOS is very poor. No it's not. In fact, it's actually very good. Vendors who keep their hardware compliant to the USB audio spec don't have to ship any driver at all. And this is how it should be on Linux, too. > Also the mpkg install file (inside the provided install dmg file) was > called something like "firmware0_and_driver.mpkg". This is why I > suspected a firmware problem. However I tried to look at the package > content and did not find any .bin file or whatever could be a standalone > firmware file. Maybe it is embedded in the driver install executable. Maybe, but they could also write the firmware permanently to the device once with that tool. That would be perfect for Linux users. Daniel

3 10

[alsa-devel] [PATCH 1/2] ASoC: pcm1792a: Remove empty capture DAI stub
by Mark Brown 23 Aug '13

23 Aug '13

From: Mark Brown <broonie(a)linaro.org> These intialisations are just what will be done for static data anyway so remove them. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1792a.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 72cf835..c57d3a5 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -162,10 +162,6 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .channels_max = 2, .rates = PCM1792A_RATES, .formats = PCM1792A_FORMATS, }, - .capture = { - .channels_min = 0, - .channels_max = 0, - }, .ops = &pcm1792a_dai_ops, }; -- 1.8.4.rc2

2 3

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Fix missing audmux external port setup check
by Andreas Pretzsch 23 Aug '13

23 Aug '13

Return value of imx_audmux_v2_configure_port(ext_port, ...) was not assigned. Following check tested the preceding call (int_port). Signed-off-by: Andreas Pretzsch <apr(a)cn-eng.de> --- Fix not tested (no matching hardware), only noticed during code reading. Patch against git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/wm8962 sound/soc/fsl/imx-wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 52a36a9..e018632 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -192,7 +192,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { -- 1.7.10.4

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.12
by Mark Brown 23 Aug '13

23 Aug '13

The following changes since commit b36f4be3de1b123d8601de062e7dbfc904f305fb: Linux 3.11-rc6 (2013-08-18 14:36:53 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.12 for you to fetch changes up to 43bcb402f84fe459102120b4e2d28d7117f16cd0: Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next (2013-08-22 14:29:02 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.12 - DAPM is now mandatory for CODEC drivers in order to avoid the repeated regressions in the special cases for non-DAPM CODECs and make it easier to integrate with other components on boards. All existing drivers have had some level of DAPM support added. - A lot of cleanups in DAPM plus support for maintaining controls in a specific state while a DAPM widget all contributed by Lars-Peter Clausen. - Core helpers for bitbanged AC'97 reset from Markus Pargmann. - New drivers and support for Analog Devices ADAU1702 and ADAU1401(a), Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson Microelectronics WM8997. - Support for building drivers that can support it cross-platform for compile test. ---------------------------------------------------------------- Axel Lin (1): ASoC: adav80x: Add module device table for adav801 Bard Liao (1): ASoC: rt5640: remove unused mux Bo Shen (7): ASoC: atmel_ssc_dai: move set dma data to startup callback ASoC: atmel_ssc_dai: add error mask define ASoC: atmel-pcm-dma: move prepare for dma to dai prepare ARM: atmel-ssc: change phybase type to dma_addr_t ASoC: atmel-pcm: use generic dmaengine framework ASoC: wm8904: fix the typo error for LINER Mux ASoC: atmel: add wm8904 based audio machine driver Charles Keepax (6): mfd: arizona: Add GPIO control register bit definitions ASoC: arizona: Add signal activity output for DRC ASoC: arizona: Add default case to silence build warning ASoC: wm5110: Correct input OSR bits for wm5110 ASoC: arizona: Add widget<->mux route into mux route macro ASoC: wm8997: Initial CODEC driver Dan Carpenter (2): ASoC: adau1701: type bug with ADAU1707_CLKDIV_UNSET ASoC: dapm: using freed pointer in dapm_kcontrol_add_widget() Daniel Mack (11): ARM: pxa: ssp: remove unnecessary warning on kzalloc() failure ARM: pxa: ssp: add shortcut for &pdev->dev ARM: pxa: ssp: add DT bindings ARM: pxa: ssp: use devm_ functions ARM: pxa: ssp: add pxa_ssp_request_of() ALSA: move dmaengine implementation from ASoC to ALSA core ASoC: pxa: pxa-ssp: add DT bindings ASoC: pxa: use snd_dmaengine_dai_dma_data ASoC: pxa: pxa-ssp: set dma filter data from startup hook ASoC: pxa: add DT bindings for pxa2xx-pcm ALSA: core: allow SND_DMAENGINE_PCM use from modules Dimitris Papastamos (4): ASoC: core: Add snd_soc_card_get_kcontrol() ASoC: wm_adsp: Simplify kcontrol handling ASoC: wm0010: Use DMA-safe memory for SPI transfers ASoC: wm_adsp: Sanitize parameter passing Fabio Estevam (11): ASoC: sglt5000: Provide the reg_stride field ASoC: fsl: fsl_ssi: Use devm_ functions ASoC: fsl: fsl_ssi: Check the return value from clk_prepare_enable() ASoC: fsl: imx-wm8962: Fix error path ASoC: wm8962: Do not call configure_bclk() inside wm8962_set_dai_sysclk() ASoC: fsl: fsl_ssi: Add MODULE_ALIAS ASoC: fsl: imx-audmux: Check the return value from clk_prepare_enable() ASoC: fsl: Fix module build ASoC: imx-mc13783: Make SND_SOC_IMX_MC13783 visible again ASoC: fsl: Drop SND_SOC_FSL_UTILS from i.mx machine code ASoC: fsl: fsl_ssi: Fix the order of resources removal Jean-Francois Moine (1): ASoC: kirkwood-i2s: fix a compilation warning Julia Lawall (5): sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument ASoC: samsung-ac97: simplify use of devm_ioremap_resource ASoC: mioa701_wm9713: Remove definition of ARRAY_AND_SIZE() ASoC: tegra20-ac97: simplify use of devm_ioremap_resource ASoC: omap: simplify platform_get_resource_byname/devm_ioremap_resource Kuninori Morimoto (13): ASoC: add ak4554 driver ASoC: ak4554: add DT support ASoC: add Renesas R-Car core feature ASoC: add Renesas R-Car module feature ASoC: add Renesas R-Car Generation feature ASoC: add Renesas R-Car SCU feature ASoC: add Renesas R-Car ADG feature ASoC: add Renesas R-Car SSI feature ASoC: rsnd: remove platform dai and add dai_id on platform setting ASoC: rsnd: add common DMAEngine method ASoC: rsnd: SSI supports DMA transfer ASoC: rsnd: SSI supports DMA transfer via BUSIF ASoC: rsnd: scu: cleanup empty functions Lars-Peter Clausen (28): ASoC: Remove unused dapm_get_snd_card() and dapm_get_soc_card() ASoC: Build adau1701 when SND_SOC_ALL_CODECS is selected ASoC: adau1701: Add adau1702 and adau1401(a) device ids ASoC: dapm: Pass snd_soc_card directly to soc_dpcm_runtime_update() ASoC: dapm: Run widget updates for shared controls at the same time ASoC: dapm: Add a update parameter to snd_soc_dapm_{mux,mixer}_update_power ASoC: tlv320aic3x: Use snd_soc_dapm_mixer_update_power ASoC: lm4857: Use table based setup for DAPM and controls ASoC: lm4857: Convert to regmap ASoC: dapm: Move snd_soc_dapm_update from dapm context to card ASoC: dapm: Pass card instead of dapm context to dapm_power_widgets() ASoC: dapm: Add a helper to get the CODEC for DAPM kcontrol ASoC: dapm: Wrap kcontrol widget list access ASoC: dapm: Move 'value' field from widget to control ASoC: dapm: Keep a list of paths per kcontrol ASoC: dapm: Make widget power register settings more flexible ASoC: dapm: Add snd_soc_dapm_add_path() helper function ASoC: dapm: Delay w->power update until the changes are written ASoC: bf5xx-ac97: Remove unused extern declaration ASoC: dapm: Check return value of snd_soc_cnew() ASoC: dapm: Fix kcontrol path list corruption ASoC: dapm: Implement mixer input auto-disable ASoC: tlv320aic26: Fix keyclick feature ASoC: mc13783: Remove embedded snd_soc_codec structs from private data structs ASoC: twl4030: Remove embedded snd_soc_codec structs from private data structs ASoC: Remove unused soc_pm_waitq ASoC: Remove unused snd_soc_info_volsw_ext() ASoC: Use snd_soc_info_enum_double() for SOC_ENUM_EXT controls Lothar Waßmann (1): ASoC: mxs: add some error messages to help identifying problems Ma Haijun (1): ASoC: wm8960: Fix ADC volume bits Marek Belisko (1): ASoC: Add PCM1681 codec driver. Mark Brown (169): ASoC: codecs: Make ALL_CODECS depend on COMPILE_TEST ASoC: pcm3008: Remove noisy version print ASoC: pcm3008: Move gpio allocation to probe ASoC: pcm3008: Check for platform data ASoC: pcm3008: Convert to devm_gpio_request_one() ASoC: wm8994: Move runtime PM init to platform device init ASoC: kirkwood: Remove unused headers ASoC: kirkwood: Enable build on non-Kirkwood platforms ASoC: kirkwood-i2s: Use devm_clk_get() for extclk ASoC: kirkwood-i2s: Remove empty remove() ASoC: kirkwood-i2s: Inline KIRKWOOD_I2S_RATES ASoC: tlv320aic3x: Add compatible strings for specific devices ASoC: tlv320aic3x: List tlv320aic3106 as a supported device ASoC: imx: Enable COMPILE_TEST builds ASoC: samsung-spdif: Convert to devm_clk_get() ASoC: omap: Enable COMPILE_TEST build for DT platforms ASoC: tegra: Remove unneeded mach-type.h incldues ASoC: tegra: Add GPIOLIB dependencies ASoC: tegra: Always use the generic dmaengine helper library ASoC: tegra: Enable COMPILE_TEST builds ASoC: imx-mc13783: Depend on ARCH_ARM ASoC: pcm: Use the power efficient workqueue for delayed powerdown ASoC: jack: Use power efficient workqueue ASoC: mxs: Enable COMPILE_TEST builds ASoC: max98090: Use power efficient workqueue ASoC: sta32x: Use power efficient workqueue ASoC: twl6040: Use power efficient workqueue ASoC: wm8350: Use power efficient workqueue ASoC: wm8753: Use power efficient workqueue ASoC: wm8994: Use power efficient workqueue ASoC: fsl_ssi: Provide register I/O functions by default ASoC: twl6040: Use power efficient workqueue ASoC: mxs: Remove unneeded mach-types.h inclusions ASoC: wm8962: Use power efficient workqueue ASoC: mxs: Depends on COMMON_CLK ASoC: dapm: Use generic power check for everything except DAIs ASoC: wm8994: Add clock bindings to the device tree ASoC: smdk_wm8994: Make driver name more unique ASoC: smdk_wm8994: Configure DAI format at init time ASoC: smdk_wm8994: Configure the MCLK1 rate based on the board Merge branch 'topic/core' of git://git.kernel.org/.../broonie/sound into asoc-dapm ASoC: samsung: Make secondary I2S DAI device a child of primary Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-new-pcm ASoC: pcm1792a: Fix build with !OF ASoC: wm8994: Fix class W controls ASoC: ad1980: Provide stub DAPM support ASoC: ads711x: Add DAPM support ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding ASoC: pcm1681: Staticise DAI driver ASoC: ad73311: Add DAPM support ASoC: ad73311: Add to list of of Analog Devices supported CODECs ASoC: max9768: Add DAPM support ASoC: cs4271: Convert to table based control init ASoC: cs4271: Add DAPM support ASoC: doc: Add documentation to MAINTAINERS patterns ASoC: compress: Use power efficient workqueue ASoC: adau1701: Use gpio_set_value_cansleep() ASoC: ak4104: Add stub DAPM support ASoC: ak4104: Manage TXE using DAPM ASoC: ak5386: Add DAPM support Merge tag 'v3.11-rc5' into asoc-dapm Merge remote-tracking branch 'asoc/fix/dapm' into asoc-dapm ASoC: dapm: Don't create routes when creating kcontrols ASoC: dapm: Ensure kcontrol list is initialised ASoC: dt: Move WM8753 pin list from Tegra board binding to CODEC binding ASoC: dt: Move RT5640 pin list from Tegra board binding to CODEC binding ASoC: dt: Move ALC5632 pin list from Tegra board binding to CODEC binding ASoC: wm8753: Fix typo in DT binding example ASoC: ak4554: Add DAPM support ASoC: max9877: Convert to use regmap API ASoC: max9877: Convert to standard CODEC driver ASoC: max9877: Add basic DAPM support Merge branch 'topic/dma' of git://git.kernel.org/.../broonie/sound into asoc-pxa ASoC: pcm1792a: Remove empty capture DAI stub ASoC: pcm1792a: Add DAPM support ASoC: pcm1681: Add DAPM support Merge branch 'topic/dma' of git://git.kernel.org/.../broonie/sound into asoc-omap Merge branch 'topic/dma' of git://git.kernel.org/.../broonie/sound into asoc-omap ASoC: samsung: Fix DTS breakage from missing dependencies ASoC: tlv320aic26: Remove direct use of internal I/O functions ASoC: tlv320aic26: Remove noisy print ASoC: tlv320aic26: Add basic DAPM support ASoC: si476x: Add DAPM support ASoC: pcm3008: Use gpio_set_value_cansleep() ASoC: pcm3008: Add DAPM support ASoC: pcm3008: Manage DAC and ADC power with DAPM ASoC: wl1273: Add stub DAPM support ASoC: cs4270: Add DAPM support ASoC: wm8727: Add DAPM support ASoC: wm8782: Add DAPM support ASoC: wm5102: Add inputs for noise and mic mixers ASoC: wm5110: Add inputs for noise and mic mixers ASoC: wm8997: Add inputs for noise and mic mixers ASoC: ac97: Provide stub DAPM integration ASoC: hdmi: Provide stub DAPM integration ASoC: bt-sco: Add generic compatible string ASoC: bt-sco: Provide stub DAPM integration ASoC: spdif: Add stub DAPM widgets for Rx ASoC: spdif: Remove duplicate const ASoC: imx: Add MODULE_LICENSE to DMA drivers ASoC: samsung: Check to see if we managed to allocate a channel ASoC: samsung: Rename DMA platform registration functions Merge remote-tracking branch 'asoc/topic/ac97' into asoc-fsl Merge remote-tracking branch 'asoc/fix/wm8960' into asoc-linus Merge remote-tracking branch 'asoc/topic/ac97' into asoc-next Merge remote-tracking branch 'asoc/topic/ad1980' into asoc-next Merge remote-tracking branch 'asoc/topic/ad73311' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adav80x' into asoc-next Merge remote-tracking branch 'asoc/topic/ads711x' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4104' into asoc-next Merge remote-tracking branch 'asoc/topic/ak4554' into asoc-next Merge remote-tracking branch 'asoc/topic/ak5386' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/au1x' into asoc-next Merge remote-tracking branch 'asoc/topic/blackfin' into asoc-next Merge remote-tracking branch 'asoc/topic/bt' into asoc-next Merge remote-tracking branch 'asoc/topic/compress' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4270' into asoc-next Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dma' into asoc-next Merge remote-tracking branch 'asoc/topic/ep93xx' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/hdmi' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/lm4857' into asoc-next Merge remote-tracking branch 'asoc/topic/max9768' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/max9877' into asoc-next Merge remote-tracking branch 'asoc/topic/mc13783' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/new-pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/nuc900' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1792a' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3008' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branch 'asoc/topic/s6000' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branch 'asoc/topic/si476x' into asoc-next Merge remote-tracking branch 'asoc/topic/spdif' into asoc-next Merge remote-tracking branch 'asoc/topic/sta32x' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic26' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/twl4030' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/txx9' into asoc-next Merge remote-tracking branch 'asoc/topic/uda134x' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wl1273' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8350' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8727' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8731' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8753' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8782' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8904' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Markus Pargmann (10): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: imx-audmux: Read default configuration from devicetree ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: imx-audmux: default configuration parser fixups ASoC: imx-audmux: Move definitions to dt-bindings ASoC: core: Generic ac97 link reset functions ASoC: fsl-ssi: ac97-slave support Michael Grzeschik (2): ASoC: fsl-ssi: imx-pcm-fiq bugfix ASoC: fsl: disable ssi irq for imx Michael Trimarchi (2): ASoC: omap-mcbsp: Support SND_SOC_DAIFMT_CBM_CFS for omap3/4 ASoC: Add PCM1792A spi mode codec support Mike Dyer (1): ASoC: wm8960: Fix PLL register writes Nicolas Ferre (1): ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Nicolin Chen (4): ASoC: fsl: Disable SSI in trigger() if RE/TE are both cleared ASoC: fsl: Set sdma peripheral type directly ASoC: spdif: Add S20_3LE and S24_LE support for dummy codec drivers ASoC: fsl: Add S/PDIF CPU DAI driver Oskar Schirmer (1): ASoC: sgtl5000: fix codec clock source transition to avoid clockless moment Padmavathi Venna (6): ASoC: Samsung: I2S: Modify driver to give more flexibility ASoC: Samsung: I2S: Add quirks as driver data in I2S ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 ARM: dts: exynos5250: move common i2s properties to exynos5 dtsi ARM: dts: Change i2s compatible string on exynos5250 ARM: dts: Change i2s compatible string on exynos5250 Peter Ujfalusi (1): ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot) Richard Fitzgerald (1): ASoC: core: init delayed_work for codec-codec links Richard Genoud (2): ASoC: wm8731: add rates constraints ASoC: atmel: update atmel SSC with DMA binding Rongjun Ying (1): ASoC: hdmi-codec: let the driver support HDMI sink Russell King (8): ASoC: uda134x: fix codec driver by converting to DAPM ASoC: spdif_transceiver: add output pin widget ASoC: kirkwood: merge struct kirkwood_dma_priv with struct kirkwood_dma_data ASoC: kirkwood: Free external clock if it is a duplicate of internal ASoC: avoid duplicated DAI routes ASoC: kirkwood: provide KIRKWOOD_PLAYCTL_ENABLE_MASK ASoC: kirkwood: combine kirkwood-i2s and kirkwood-dma drivers ASoC: kirkwood: move calculation of max buffer size to kirkwood.h Sachin Kamat (2): ASoC: s6105-ipcam: Fix incorrect placement of __initdata ASoC: pxa: Remove duplicate inclusion of dmaengine.h Steffen Trumtrar (1): ASoC: fsl-ssi: add SSIEN errata work around Stephen Warren (2): ASoC: tegra: fix compile warning in AC'97 driver ASoC: tegra: support a Mic Jack in the Tegra+RT5640 machine driver Tushar Behera (1): ASoC: samsung: Fix build error with dma function rename Vinod Koul (1): ASoC: compress: use soc_xxx handlers for metadata Wei Yongjun (7): ASoC: db1200: add .owner to struct snd_soc_card ASoC: imx_mc13783: add .owner to struct snd_soc_card ASoC: brownstone: add .owner to struct snd_soc_card ASoC: ttc_dkb: add .owner to struct snd_soc_card ASoC: mop500: add .owner to struct snd_soc_card ASoC: imx-sgtl5000: fix error return code in imx_sgtl5000_probe() ASoC: rcar: fix return value check in rsnd_gen1_probe() Wolfram Sang (5): ASoC: au1x: don't check resource with devm_ioremap_resource ASoC: ep93xx: don't check resource with devm_ioremap_resource ASoC: nuc900: don't check resource with devm_ioremap_resource ASoC: pxa: don't check resource with devm_ioremap_resource ASoC: txx9: don't check resource with devm_ioremap_resource .../devicetree/bindings/misc/atmel-ssc.txt | 23 +- .../devicetree/bindings/serial/mrvl,pxa-ssp.txt | 65 + Documentation/devicetree/bindings/sound/ak4554.c | 11 + .../devicetree/bindings/sound/alc5632.txt | 19 + .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 + .../devicetree/bindings/sound/atmel-wm8904.txt | 55 + .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 12 + .../devicetree/bindings/sound/imx-audmux.txt | 9 + .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 + .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 24 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 26 +- .../bindings/sound/nvidia,tegra-audio-wm8753.txt | 27 +- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 24 +- .../devicetree/bindings/sound/pcm1792a.txt | 18 + Documentation/devicetree/bindings/sound/rt5640.txt | 20 + .../devicetree/bindings/sound/samsung-i2s.txt | 22 +- .../devicetree/bindings/sound/soc-ac97link.txt | 28 + .../devicetree/bindings/sound/ti,pcm1681.txt | 15 + .../devicetree/bindings/sound/tlv320aic3x.txt | 9 +- Documentation/devicetree/bindings/sound/wm8731.txt | 9 + Documentation/devicetree/bindings/sound/wm8753.txt | 24 +- Documentation/devicetree/bindings/sound/wm8903.txt | 19 + Documentation/devicetree/bindings/sound/wm8994.txt | 4 + MAINTAINERS | 2 + arch/arm/boot/dts/exynos5250.dtsi | 9 +- arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 +- arch/arm/plat-pxa/ssp.c | 171 ++- include/dt-bindings/sound/fsl-imx-audmux.h | 56 + include/linux/atmel-ssc.h | 2 +- include/linux/mfd/arizona/gpio.h | 96 ++ include/linux/platform_data/asoc-s3c.h | 1 + include/linux/platform_data/omap-abe-twl6040.h | 49 - include/linux/pxa2xx_ssp.h | 11 + include/sound/pxa2xx-lib.h | 7 - include/sound/rcar_snd.h | 84 ++ include/sound/soc-dapm.h | 199 ++-- include/sound/soc-dpcm.h | 2 +- include/sound/soc.h | 39 +- sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 +- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + .../soc-dmaengine-pcm.c => core/pcm_dmaengine.c} | 0 sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/atmel/Kconfig | 21 + sound/soc/atmel/Makefile | 4 + sound/soc/atmel/atmel-pcm-dma.c | 118 +- sound/soc/atmel/atmel_ssc_dai.c | 36 +- sound/soc/atmel/atmel_wm8904.c | 254 ++++ sound/soc/atmel/sam9x5_wm8731.c | 208 ++++ sound/soc/au1x/db1200.c | 4 + sound/soc/au1x/psc-ac97.c | 3 - sound/soc/blackfin/bf5xx-ac97.h | 1 - sound/soc/cirrus/ep93xx-ac97.c | 3 - sound/soc/cirrus/ep93xx-i2s.c | 3 - sound/soc/codecs/Kconfig | 19 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/ac97.c | 15 + sound/soc/codecs/ad1980.c | 43 + sound/soc/codecs/ad73311.c | 22 +- sound/soc/codecs/adau1701.c | 25 +- sound/soc/codecs/adav80x.c | 13 +- sound/soc/codecs/ads117x.c | 29 +- sound/soc/codecs/ak4104.c | 34 +- sound/soc/codecs/ak4554.c | 106 ++ sound/soc/codecs/ak5386.c | 17 +- sound/soc/codecs/arizona.c | 69 +- sound/soc/codecs/arizona.h | 5 +- sound/soc/codecs/bt-sco.c | 22 +- sound/soc/codecs/cs4270.c | 20 + sound/soc/codecs/cs4271.c | 30 +- sound/soc/codecs/hdmi.c | 30 +- sound/soc/codecs/lm4857.c | 107 +- sound/soc/codecs/max9768.c | 16 + sound/soc/codecs/max98090.c | 10 +- sound/soc/codecs/max9877.c | 294 ++--- sound/soc/codecs/mc13783.c | 1 - sound/soc/codecs/pcm1681.c | 339 ++++++ sound/soc/codecs/pcm1792a.c | 257 ++++ sound/soc/codecs/pcm1792a.h | 26 + sound/soc/codecs/pcm3008.c | 150 ++- sound/soc/codecs/rt5640.c | 26 - sound/soc/codecs/sgtl5000.c | 10 +- sound/soc/codecs/si476x.c | 14 + sound/soc/codecs/spdif_receiver.c | 17 +- sound/soc/codecs/spdif_transmitter.c | 18 +- sound/soc/codecs/sta32x.c | 10 +- sound/soc/codecs/tlv320aic26.c | 51 +- sound/soc/codecs/tlv320aic3x.c | 56 +- sound/soc/codecs/twl4030.c | 2 - sound/soc/codecs/twl6040.c | 7 +- sound/soc/codecs/uda134x.c | 88 +- sound/soc/codecs/wl1273.c | 17 + sound/soc/codecs/wm0010.c | 12 +- sound/soc/codecs/wm5102.c | 53 +- sound/soc/codecs/wm5110.c | 35 +- sound/soc/codecs/wm8350.c | 6 +- sound/soc/codecs/wm8727.c | 17 +- sound/soc/codecs/wm8731.c | 60 +- sound/soc/codecs/wm8753.c | 5 +- sound/soc/codecs/wm8782.c | 17 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8960.c | 10 +- sound/soc/codecs/wm8962.c | 7 +- sound/soc/codecs/wm8994.c | 39 +- sound/soc/codecs/wm8995.c | 5 +- sound/soc/codecs/wm8997.c | 1175 +++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + sound/soc/codecs/wm_adsp.c | 124 +- sound/soc/codecs/wm_adsp.h | 3 +- sound/soc/codecs/wm_hubs.c | 8 +- sound/soc/fsl/Kconfig | 13 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ sound/soc/fsl/fsl_ssi.c | 500 ++++++-- sound/soc/fsl/imx-audmux.c | 78 +- sound/soc/fsl/imx-audmux.h | 52 +- sound/soc/fsl/imx-mc13783.c | 1 + sound/soc/fsl/imx-pcm-dma.c | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 20 +- sound/soc/fsl/imx-pcm.h | 26 +- sound/soc/fsl/imx-sgtl5000.c | 4 +- sound/soc/fsl/imx-ssi.c | 11 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/imx-wm8962.c | 3 +- sound/soc/kirkwood/Kconfig | 11 +- sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 108 +- sound/soc/kirkwood/kirkwood-i2s.c | 67 +- sound/soc/kirkwood/kirkwood-openrd.c | 6 +- sound/soc/kirkwood/kirkwood-t5325.c | 6 +- sound/soc/kirkwood/kirkwood.h | 11 +- sound/soc/mxs/Kconfig | 3 +- sound/soc/mxs/mxs-saif.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 30 +- sound/soc/nuc900/nuc900-ac97.c | 3 - sound/soc/omap/Kconfig | 8 +- sound/soc/omap/omap-abe-twl6040.c | 133 +-- sound/soc/omap/omap-dmic.c | 9 +- sound/soc/omap/omap-mcbsp.c | 5 + sound/soc/omap/omap-mcpdm.c | 3 - sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 7 +- sound/soc/pxa/mmp-pcm.c | 7 +- sound/soc/pxa/mmp-sspa.c | 15 +- sound/soc/pxa/pxa-ssp.c | 76 +- sound/soc/pxa/pxa2xx-ac97.c | 67 +- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/pxa/ttc-dkb.c | 1 + sound/soc/s6000/s6105-ipcam.c | 2 +- sound/soc/samsung/ac97.c | 11 +- sound/soc/samsung/dma.c | 12 +- sound/soc/samsung/dma.h | 4 +- sound/soc/samsung/i2s-regs.h | 51 +- sound/soc/samsung/i2s.c | 193 ++- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/smdk_wm8994.c | 58 +- sound/soc/samsung/spdif.c | 12 +- sound/soc/sh/Kconfig | 7 + sound/soc/sh/Makefile | 3 + sound/soc/sh/rcar/Makefile | 2 + sound/soc/sh/rcar/adg.c | 234 ++++ sound/soc/sh/rcar/core.c | 861 ++++++++++++++ sound/soc/sh/rcar/gen.c | 280 +++++ sound/soc/sh/rcar/rsnd.h | 302 +++++ sound/soc/sh/rcar/scu.c | 236 ++++ sound/soc/sh/rcar/ssi.c | 728 ++++++++++++ sound/soc/soc-compress.c | 13 +- sound/soc/soc-core.c | 236 +++- sound/soc/soc-dapm.c | 810 +++++++------ sound/soc/soc-jack.c | 2 +- sound/soc/soc-pcm.c | 15 +- sound/soc/spear/Kconfig | 2 +- sound/soc/tegra/Kconfig | 14 +- sound/soc/tegra/tegra20_ac97.c | 8 - sound/soc/tegra/tegra_alc5632.c | 2 - sound/soc/tegra/tegra_rt5640.c | 1 + sound/soc/tegra/tegra_wm8753.c | 2 - sound/soc/tegra/trimslice.c | 2 - sound/soc/txx9/txx9aclc-ac97.c | 3 - sound/soc/ux500/mop500.c | 1 + 193 files changed, 10275 insertions(+), 2357 deletions(-) create mode 100644 Documentation/devicetree/bindings/serial/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4554.c create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (87%) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1681.txt create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h create mode 100644 include/linux/mfd/arizona/gpio.h delete mode 100644 include/linux/platform_data/omap-abe-twl6040.h create mode 100644 include/sound/rcar_snd.h rename sound/{soc/soc-dmaengine-pcm.c => core/pcm_dmaengine.c} (100%) create mode 100644 sound/soc/atmel/atmel_wm8904.c create mode 100644 sound/soc/atmel/sam9x5_wm8731.c create mode 100644 sound/soc/codecs/ak4554.c create mode 100644 sound/soc/codecs/pcm1681.c create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/sh/rcar/Makefile create mode 100644 sound/soc/sh/rcar/adg.c create mode 100644 sound/soc/sh/rcar/core.c create mode 100644 sound/soc/sh/rcar/gen.c create mode 100644 sound/soc/sh/rcar/rsnd.h create mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/ssi.c

2 1

Re: [alsa-devel] oops during boot with CONFIG_SND_DYNAMIC_MINORS not set
by Takashi Iwai 23 Aug '13

23 Aug '13

At Fri, 23 Aug 2013 00:56:01 +0300, Stratos Karafotis wrote: > > On 08/23/2013 12:23 AM, Takashi Iwai wrote: > > At Thu, 22 Aug 2013 19:03:44 +0300, > > Stratos Karafotis wrote: > >> > >> On 08/22/2013 10:59 AM, Takashi Iwai wrote: > >>> At Thu, 22 Aug 2013 00:42:41 +0300, > >>> Stratos Karafotis wrote: > >>>> > >>>> Hi, > >>>> > >>>> I get the following oops during boot when build with CONFIG_SND_DYNAMIC_MINORS > >>>> not set (3.11-rc6). > >>>> The issue is vanished building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > >>>> as suggested in printk message. > >>>> > >>>> Regards, > >>>> Stratos > >>>> > >>>> > >>> > >>> Could you check the patch below? > >>> Thanks! > >>> > >>> > >>> Takashi > >>> > >>> --- > >>> From: Takashi Iwai <tiwai(a)suse.de> > >>> Subject: [PATCH] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n > >>> > >>> Without the dynamic minor assignment, HDMI codec may have less PCM > >>> instances than the number of pins, which eventually leads to Oops. > >>> > >>> Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> > >>> Cc: <stable(a)vger.kernel.org> > >>> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > >>> --- > >>> sound/pci/hda/patch_hdmi.c | 3 +++ > >>> 1 file changed, 3 insertions(+) > >>> > >>> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > >>> index 030ca86..e2cb92b 100644 > >>> --- a/sound/pci/hda/patch_hdmi.c > >>> +++ b/sound/pci/hda/patch_hdmi.c > >>> @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) > >>> struct snd_pcm_chmap *chmap; > >>> struct snd_kcontrol *kctl; > >>> int i; > >>> + > >>> + if (pin_idx >= codec->num_pcms) > >>> + break; > >>> err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, > >>> SNDRV_PCM_STREAM_PLAYBACK, > >>> NULL, 0, pin_idx, &chmap); > >>> > >> > >> Hi, > >> > >> Unfortunately, still the same problem after applying your patch. > > > > Bah, it's a wrong one. The patch below should work better. > > Sorry for inconvenience. > > No problem! :) > > Yes, no more oops with the latest patch. > Tested with CONFIG_SND_DYNAMIC_MINORS=y and n. > > There is an error -16 now (please see below), but I guess it's normal(?) Yes, it's the expected behavior. Thanks for quick tests! Takashi > Thanks, > Stratos > > > [ 16.392453] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 16.392457] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 16.392460] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 16.392461] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 16.409640] ALSA sound/core/control.c:349 control 0:0:0:HDMI/DP Jack:0 is already present > [ 16.410627] hda_codec: cannot build controls for #3 (error -16) > [ 16.411683] input: HDA NVidia HDMI/DP as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input15 > [ 16.411844] input: HDA NVidia HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input16 > [ 16.411986] input: HDA NVidia HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:01.0/0000:01:00.1/sound/card1/input17 > [ 19.682495] EXT4-fs (sda2): mounted filesystem with ordered data mode. Opts: (null) > [ 21.367420] FAT-fs (sda1): Volume was not properly unmounted. Some data may be corrupt. Please run fsck. > [ 21.420191] EXT4-fs (dm-2): mounted filesystem with ordered data mode. Opts: (null) > [ 21.847085] type=1305 audit(1377207061.920:3): audit_pid=534 old=0 auid=4294967295 ses=4294967295 > res=1 > [ 22.164707] alsactl[549]: segfault at 1 ip 00007f6c8ad58a7d sp 00007fff19912de0 error 4 in libc-2.17.so[7f6c8ad10000+1b6000] >

1 0

[alsa-devel] [RFC PATCH 3/3] ALSA: hda - Haswell codec exposes device list/select info on pins
by mengdong.lin＠intel.com 23 Aug '13

23 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is only to allow codec proc file to expose devices list/select info for Haswell codec pins. Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list is empty, meaning no pin is multi-streaming capaple. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..87ca984 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,8 +1989,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); -- 1.8.1.2

1 0

[alsa-devel] [RFC PATCH 2/3] ALSA: hda - add device list & select info of display pins to codec proc file
by mengdong.lin＠intel.com 23 Aug '13

23 Aug '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If a display codec supports multi-stream transport on the pins, the pin's device list length and device entries will be exposed to codec proc file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 9760f00..fd62d8f 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +582,38 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + union hda_devlist_entry dev_list[AC_MAX_DEVLIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEVLIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %ld, ELDV = %ld, IA = %ld\n", + i, dev_list[i].bits.present_detect, + dev_list[i].bits.eld_valid, dev_list[i].bits.inactive); + } + + return 0; + +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -751,6 +783,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); -- 1.8.1.2

1 0

Re: [alsa-devel] [PATCH] rme96 Add missing vmalloc.h inclusion
by Takashi Iwai 22 Aug '13

22 Aug '13

At Thu, 22 Aug 2013 15:06:38 +0200, Knut Petersen wrote: > > [1 <text/plain; ISO-8859-1 (7bit)>] > On 22.08.2013 10:53, Takashi Iwai wrote: > > Actually this triggers the double free. It's freed in > > snd_rme96_free() via snd_card_free(). > Yes. Now I see that. > > I applied the patch with removal of this double vfree call. > Thanks. > > Unfortunately there still is a problem with the code: > vmalloc() and vfree() were used without inclusion > of linux/vmalloc.h. Indeed. Applied now. Thanks. Takashi > > cu, > Knut > > > [2 0001-alsa-rme96-Add-missing-inclusion-of-linux-vmalloc.h.patch <text/x-patch (7bit)>] > >From 2a3f98ff69308f3c2ee508ac0887e1f8bf493d64 Mon Sep 17 00:00:00 2001 > From: Knut Petersen <Knut_Petersen(a)t-online.de> > Date: Thu, 22 Aug 2013 14:36:16 +0200 > Subject: [PATCH] alsa/rme96: Add missing inclusion of linux/vmalloc.h > > Signed-off-by: Knut Petersen <Knut_Petersen(a)t-online.de> > --- > sound/pci/rme96.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c > index 2acf565..cc6deca 100644 > --- a/sound/pci/rme96.c > +++ b/sound/pci/rme96.c > @@ -28,6 +28,7 @@ > #include <linux/interrupt.h> > #include <linux/pci.h> > #include <linux/module.h> > +#include <linux/vmalloc.h> > > #include <sound/core.h> > #include <sound/info.h> > -- > 1.8.1.4 >

1 0

Re: [alsa-devel] oops during boot with CONFIG_SND_DYNAMIC_MINORS not set
by Takashi Iwai 22 Aug '13

22 Aug '13

At Thu, 22 Aug 2013 19:03:44 +0300, Stratos Karafotis wrote: > > On 08/22/2013 10:59 AM, Takashi Iwai wrote: > > At Thu, 22 Aug 2013 00:42:41 +0300, > > Stratos Karafotis wrote: > >> > >> Hi, > >> > >> I get the following oops during boot when build with CONFIG_SND_DYNAMIC_MINORS > >> not set (3.11-rc6). > >> The issue is vanished building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > >> as suggested in printk message. > >> > >> Regards, > >> Stratos > >> > >> > > > > Could you check the patch below? > > Thanks! > > > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n > > > > Without the dynamic minor assignment, HDMI codec may have less PCM > > instances than the number of pins, which eventually leads to Oops. > > > > Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> > > Cc: <stable(a)vger.kernel.org> > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/patch_hdmi.c | 3 +++ > > 1 file changed, 3 insertions(+) > > > > diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c > > index 030ca86..e2cb92b 100644 > > --- a/sound/pci/hda/patch_hdmi.c > > +++ b/sound/pci/hda/patch_hdmi.c > > @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) > > struct snd_pcm_chmap *chmap; > > struct snd_kcontrol *kctl; > > int i; > > + > > + if (pin_idx >= codec->num_pcms) > > + break; > > err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, > > SNDRV_PCM_STREAM_PLAYBACK, > > NULL, 0, pin_idx, &chmap); > > > > Hi, > > Unfortunately, still the same problem after applying your patch. Bah, it's a wrong one. The patch below should work better. Sorry for inconvenience. Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH v2] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n Without the dynamic minor assignment, HDMI codec may have less PCM instances than the number of pins, which eventually leads to Oops. Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..9f35862 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; + + if (!codec->pcm_info[pin_idx].pcm) + break; err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); -- 1.8.3.4

1 0

[alsa-devel] [PATCH v2 00/10] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 22 Aug '13

22 Aug '13

Hi, This patch series attempts to fix a long starting problem with the concurrent usage of playback and capture on implicit feedback devices. Applies against current for-next (HEAD b43dd416be21bc8ad60984e13def032f01aaaa18 ). v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065517.html Thanks to Clemens I was able to get rid of the get_interface workaround, so I believe it's ready now, unless there's any other issues. Other changes: * Cleaned up and moved the set_param flag patch to #7. * Fixed a NULL dereference in patch #8 for explicit feedback. * Fixed a logic error in patch #9. * Added patch #10. Cheers, Eldad Eldad Zack (10): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: correct ep use_count semantics (add set_param flag) ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: remove altset_idx from snd_usb_substream sound/usb/card.h | 2 +- sound/usb/endpoint.c | 76 +++++++++++++++++------- sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 161 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/pcm.h | 2 + sound/usb/proc.c | 4 +- 6 files changed, 192 insertions(+), 64 deletions(-) -- 1.8.1.5

3 17

[alsa-devel] [PATCH v10 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 22 Aug '13

22 Aug '13

Changelog: v9->v10: * Removed "dummy" words from binding doc in PATCH-2. v8->v9: * Use bool instead of atomic_t. * Use clk_prepare_enable() instead. * Dropped dumpregs(). * Dropped unnecessary clk_round_rate(). * Dropped unused clock source enum. * Revised dev_err() message. v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ sound/soc/fsl/imx-spdif.c | 134 +++ 7 files changed, 1662 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 10

[alsa-devel] [PATCH v5 1/1] ASoc: kirkwood: add DT support to the mvebu audio subsystem
by Jean-Francois Moine 22 Aug '13

22 Aug '13

This patch adds DT support to the audio subsystem of the mvebu family (Kirkwood, Dove, Armada 370). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- This patch does not add any new function as S/PDIF or DPCM support. --- .../devicetree/bindings/sound/mvebu-audio.txt | 29 ++++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 26 +++++++++++++++++----- 2 files changed, 49 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 0000000..7e5fd37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) audio controller + +Required properties: + +- compatible: "marvell,mvebu-audio" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "extclk" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-audio"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c.next b/sound/soc/kirkwood/kirkwood-i2s.c index e5f3f7a..a4170b4 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c.next +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -22,6 +22,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "mvebu-audio" @@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); return -EINVAL; } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); return PTR_ERR(priv->clk); @@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id mvebu_audio_of_match[] = { + { .compatible = "marvell,mvebu-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mvebu_audio_of_match), }, }; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

2 1

[alsa-devel] [PATCH] ASoC: samsung: Fix build error with dma function rename
by Tushar Behera 22 Aug '13

22 Aug '13

commit 85ff3c29d720 ("ASoC: samsung: Rename DMA platform registration functions") renames the DMA registration functions. Fix the places where it was left out. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 1566afe..e54256f 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -594,7 +594,7 @@ static int s3c_pcm_dev_probe(struct platform_device *pdev) goto err5; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -623,7 +623,7 @@ static int s3c_pcm_dev_remove(struct platform_device *pdev) struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); pm_runtime_disable(&pdev->dev); diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 47e2386..ea885cb 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -176,7 +176,7 @@ static int s3c2412_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the DMA: %d\n", ret); goto err; @@ -190,7 +190,7 @@ err: static int s3c2412_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 8b34145..9c8ebd8 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -480,7 +480,7 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) return ret; } - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { pr_err("failed to register the dma: %d\n", ret); goto err; @@ -494,7 +494,7 @@ err: static int s3c24xx_iis_dev_remove(struct platform_device *pdev) { - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 5ea70ab..28487dc 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -442,7 +442,7 @@ static int spdif_probe(struct platform_device *pdev) spdif->dma_playback = &spdif_stereo_out; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to register DMA: %d\n", ret); goto err5; @@ -468,7 +468,7 @@ static int spdif_remove(struct platform_device *pdev) struct samsung_spdif_info *spdif = &spdif_info; struct resource *mem_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); iounmap(spdif->regs); -- 1.7.9.5

2 1

[alsa-devel] ice1712 suspend/resume for M-Audio Audiophile 2496
by Tommi Uimonen 22 Aug '13

22 Aug '13

Hello, the lack of suspend/resume support for ice1712 driver has been bugging me a lot, and I was wondering if it's possible to implement it by following how it's done in 1724.c? I don't really know much about the driver, but after some comparison between 1712.c and 1724.c it seems doable. At least the suspend part seems straightforward, just changing ICEMT1724 with ICEMT and ICEREG1724 with ICEREG. To resume, I should add some new functions like snd_ice1712_chip_reset (I guess it follows the same logic as in 1724.c, just use ICE1712_RESET instead) and maybe some spdif stuff but I think the 1724 has different spdif options. I don't have any 1724 cards so can't be sure, but 1712 does not have spdif rate. Maybe it will work without those, at least for me since I don't use spdif. And then add this to the pci_driver ice1712_driver struct: .driver = { .pm = SND_ICE1712_PM_OPS, } and define the SIMPLE_DEV_PM_OPS accordingly. Or is this already under work by someone? Tommi

1 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Fix ALC283 headphone pop-noise better
by Takashi Iwai 22 Aug '13

22 Aug '13

From: Kailang Yang <kailang(a)realtek.com> Fixed ALC283 D3 to D0 and D0 to D3 Headphone pop noise. The previous fix [c5177c86: ALSA: hda - Fix the noise after suspend on ALC283 codec] doesn't work sufficiently for some laptops. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 111 +++++++++++++++++++++++++++++------------- 1 file changed, 76 insertions(+), 35 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d00639..4bdccd1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2527,6 +2527,7 @@ enum { ALC269_TYPE_ALC269VD, ALC269_TYPE_ALC280, ALC269_TYPE_ALC282, + ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, }; @@ -2552,6 +2553,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC269VB: case ALC269_TYPE_ALC269VD: case ALC269_TYPE_ALC282: + case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: ssids = alc269_ssids; break; @@ -2586,6 +2588,74 @@ static void alc269_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc283_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) + return; + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ + /* Headphone capless set to high power mode */ + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + if (hp_pin_sense) + msleep(85); + /* Index 0x46 Combo jack auto switch control 2 */ + /* 3k pull low control for Headset jack. */ + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val & ~(3 << 12)); + /* Headphone capless set to normal mode */ + alc_write_coef_idx(codec, 0x43, 0x9614); +} + +static void alc283_shutup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) { + alc269_shutup(codec); + return; + } + + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val | (3 << 12)); + + if (hp_pin_sense) + msleep(85); + snd_hda_shutup_pins(codec); + alc_write_coef_idx(codec, 0x43, 0x9614); +} + static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, unsigned int val) { @@ -2715,12 +2785,6 @@ static int alc269_resume(struct hda_codec *codec) if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); - /* clear the power-save mode for ALC283 */ - if (codec->vendor_id == 0x10ec0283) { - alc_write_coef_idx(codec, 0x4, 0xaf01); - alc_write_coef_idx(codec, 0x6, 0x2104); - } - return 0; } #endif /* CONFIG_PM */ @@ -3815,30 +3879,6 @@ static void alc269_fill_coef(struct hda_codec *codec) alc_write_coef_idx(codec, 0x4, val | (1<<11)); } -/* don't clear mic pin; otherwise it results in noise in D3 */ -static void alc283_headset_shutup(struct hda_codec *codec) -{ - int i; - - if (codec->bus->shutdown) - return; - - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); - /* use read here for syncing after issuing each verb */ - if (pin->nid != 0x19) - snd_hda_codec_read(codec, pin->nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } - - alc_write_coef_idx(codec, 0x4, 0x0f01); /* power save */ - alc_write_coef_idx(codec, 0x6, 0x2100); /* power save */ - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_VREFHIZ); - codec->pins_shutup = 1; -} - /* */ static int patch_alc269(struct hda_codec *codec) @@ -3853,9 +3893,6 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; - if (codec->vendor_id == 0x10ec0283) - spec->shutup = alc283_headset_shutup; - snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -3897,11 +3934,15 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0290: spec->codec_variant = ALC269_TYPE_ALC280; break; - case 0x10ec0233: case 0x10ec0282: - case 0x10ec0283: spec->codec_variant = ALC269_TYPE_ALC282; break; + case 0x10ec0233: + case 0x10ec0283: + spec->codec_variant = ALC269_TYPE_ALC283; + spec->shutup = alc283_shutup; + spec->init_hook = alc283_init; + break; case 0x10ec0284: case 0x10ec0292: spec->codec_variant = ALC269_TYPE_ALC284; -- 1.8.3.4

2 3

Re: [alsa-devel] [PATCH -next] ASoC: mmp-pcm: remove duplicated include from mmp-pcm.c
by Mark Brown 22 Aug '13

22 Aug '13

On Thu, Aug 22, 2013 at 10:37:20AM +0800, Wei Yongjun wrote: > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Remove duplicated include. Sorry, I already applied a patch for this which I saw first (though yours was actually sent sooner). Thanks.

1 0

[alsa-devel] [PATCH 1/1] ASoC: pxa: Remove duplicate inclusion of dmaengine.h
by Sachin Kamat 22 Aug '13

22 Aug '13

dmaengine.h header file was included twice. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/pxa/mmp-pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 9a97843..8235e23 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -17,7 +17,6 @@ #include <linux/dmaengine.h> #include <linux/platform_data/dma-mmp_tdma.h> #include <linux/platform_data/mmp_audio.h> -#include <linux/dmaengine.h> #include <sound/pxa2xx-lib.h> #include <sound/core.h> -- 1.7.9.5

2 1

[alsa-devel] [PATCH] rme96 synchronization support
by Knut Petersen 22 Aug '13

22 Aug '13

Hi everybody! As I need support for synchronized pcm streams on an RME Digi96/8 PAD I wrote the attached patch. It does work here. My own program, trying to get synchronized capture/playback streams, succeeds. Programs that use either the playback or the capture part of the hardware are not affected by start / pause/ stop operations of applications using the other part of the hardware (tested with aplay, arecord, and ecasound). I wrote the test code and the patch, so it is not impossible that I acted on wrong assumptions in both places. Please have a close look at the patch. cu, knut

3 15

[alsa-devel] [PATCH] Update iatomic.h functions definitions for mips
by jiashuo.kang＠gmail.com 22 Aug '13

22 Aug '13

From: Kai Kang <jiashuo.kang at gmail.com> Functions atomic_add(s) and atomic_sub(s) are defined with 'extern __inline__' that may cause compile fails when cross compile for mips. The error message looks like: | pcm/.libs/libpcm.a(pcm_meter.o): In function `snd_pcm_meter_update_scope': | .../alsa-lib-1.0.27.2/src/pcm/pcm_meter.c:139: undefined reference to `atomic_sub' Replace the 'extern __inline__' with 'static __inline__' to fix this issue. Signed-off-by: Kai Kang <jiashuo.kang at gmail.com> --- include/iatomic.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/iatomic.h b/include/iatomic.h index 5711fe8..2393297 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -720,7 +720,7 @@ typedef struct { volatile int counter; } atomic_t; * Atomically adds @i to @v. Note that the guaranteed useful range * of an atomic_t is only 24 bits. */ -extern __inline__ void atomic_add(int i, atomic_t * v) +static __inline__ void atomic_add(int i, atomic_t * v) { unsigned long temp; @@ -744,7 +744,7 @@ extern __inline__ void atomic_add(int i, atomic_t * v) * Atomically subtracts @i from @v. Note that the guaranteed * useful range of an atomic_t is only 24 bits. */ -extern __inline__ void atomic_sub(int i, atomic_t * v) +static __inline__ void atomic_sub(int i, atomic_t * v) { unsigned long temp; @@ -763,7 +763,7 @@ extern __inline__ void atomic_sub(int i, atomic_t * v) /* * Same as above, but return the result value */ -extern __inline__ int atomic_add_return(int i, atomic_t * v) +static __inline__ int atomic_add_return(int i, atomic_t * v) { unsigned long temp, result; @@ -784,7 +784,7 @@ extern __inline__ int atomic_add_return(int i, atomic_t * v) return result; } -extern __inline__ int atomic_sub_return(int i, atomic_t * v) +static __inline__ int atomic_sub_return(int i, atomic_t * v) { unsigned long temp, result; -- 1.8.1.2

3 2

Re: [alsa-devel] oops during boot with CONFIG_SND_DYNAMIC_MINORS not set
by Takashi Iwai 22 Aug '13

22 Aug '13

At Thu, 22 Aug 2013 00:42:41 +0300, Stratos Karafotis wrote: > > Hi, > > I get the following oops during boot when build with CONFIG_SND_DYNAMIC_MINORS > not set (3.11-rc6). > The issue is vanished building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > as suggested in printk message. > > Regards, > Stratos > > > [ 8.670497] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 8.670500] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 8.670501] ALSA sound/pci/hda/hda_codec.c:4506 Too many HDMI devices > [ 8.670502] ALSA sound/pci/hda/hda_codec.c:4508 Consider building the kernel with CONFIG_SND_DYNAMIC_MINORS=y > [ 8.688015] BUG: unable to handle kernel NULL pointer dereference at 0000000000000018 > [ 8.688609] IP: [<ffffffffa0086992>] snd_pcm_add_chmap_ctls+0xd2/0x160 [snd_pcm] > [ 8.689191] PGD 0 > [ 8.689762] Oops: 0000 [#1] SMP > [ 8.690326] Modules linked in: snd_hda_codec_hdmi snd_hda_codec_realtek arc4 rt2800pci eeprom_93cx6 rt2x00pci rt2800lib crc_ccitt rt2x00mmio rt2x00lib mac80211 cfg80211 eeepc_wmi asus_wmi sparse_keymap rfkill snd_hda_intel(+) snd_hda_codec snd_hwdep snd_seq iTCO_wdt iTCO_vendor_support r8169 mii i2c_i801 i2c_core snd_seq_device snd_pcm serio_raw pcspkr lpc_ich mfd_core snd_page_alloc snd_timer snd soundcore binfmt_misc uinput usb_storage wmi video > [ 8.692808] CPU: 1 PID: 417 Comm: systemd-udevd Not tainted 3.11.0-rc6 #3 > [ 8.693424] Hardware name: ASUSTeK COMPUTER INC. CM6870/CM6870, BIOS 0606 08/27/2012 > [ 8.694044] task: ffff880210091750 ti: ffff880210598000 task.ti: ffff880210598000 > [ 8.694666] RIP: 0010:[<ffffffffa0086992>] [<ffffffffa0086992>] snd_pcm_add_chmap_ctls+0xd2/0x160 [snd_pcm] > [ 8.695310] RSP: 0018:ffff880210599968 EFLAGS: 00010246 > [ 8.695954] RAX: ffffffffa008c51e RBX: ffff880212d20b80 RCX: 0000000000000000 > [ 8.696607] RDX: ffffffffa008c533 RSI: ffff880212d20b80 RDI: ffff880210599980 > [ 8.697261] RBP: ffff8802105999f8 R08: 0000000000000000 R09: ffff880216003b00 > [ 8.697923] R10: 0000000000000000 R11: ffff8802133d93c0 R12: 0000000000000000 > [ 8.698585] R13: ffff880210599a10 R14: 0000000000000000 R15: ffff880210599980 > [ 8.699255] FS: 00007fe8eb8ea880(0000) GS:ffff88021ec40000(0000) knlGS:0000000000000000 > [ 8.699936] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 > [ 8.700619] CR2: 0000000000000018 CR3: 00000002101c2000 CR4: 00000000001407e0 > [ 8.701310] Stack: > [ 8.702005] 0000000000000000 000000000b5a4000 0000000000000000 0000000000000003 > [ 8.702715] 0000000000000000 ffffffffa008c51e 1000001100000000 0000000000000000 > [ 8.703422] ffffffffa0084a80 ffffffffa0086cd0 0000000000000000 ffffffffa0085b30 > [ 8.704132] Call Trace: > [ 8.704836] [<ffffffffa0084a80>] ? snd_pcm_hw_rule_msbits+0x50/0x50 [snd_pcm] > [ 8.705558] [<ffffffffa0086cd0>] ? snd_pcm_hw_rule_ratdens+0x2b0/0x2b0 [snd_pcm] > [ 8.706281] [<ffffffffa0085b30>] ? snd_pcm_hw_param_last+0x240/0x240 [snd_pcm] > [ 8.707008] [<ffffffffa02af7ed>] generic_hdmi_build_controls+0x14d/0x1e0 [snd_hda_codec_hdmi] > [ 8.707748] [<ffffffffa02ae827>] ? generic_hdmi_init+0xb7/0xd0 [snd_hda_codec_hdmi] > [ 8.708493] [<ffffffffa013a812>] snd_hda_codec_build_controls+0x1c2/0x220 [snd_hda_codec] > [ 8.709242] [<ffffffffa0135255>] ? snd_hda_codec_configure+0x295/0x450 [snd_hda_codec] > [ 8.709990] [<ffffffffa013a898>] snd_hda_build_controls+0x28/0x80 [snd_hda_codec] > [ 8.710728] [<ffffffffa00fbbed>] azx_probe_continue+0x84d/0xcc0 [snd_hda_intel] > [ 8.711456] [<ffffffffa00fb060>] ? perf_trace_azx_pcm_trigger+0xe0/0xe0 [snd_hda_intel] > [ 8.712187] [<ffffffffa00f9ee0>] ? azx_resume+0x130/0x130 [snd_hda_intel] > [ 8.712916] [<ffffffffa00fac20>] ? azx_pcm_prepare+0x5f0/0x5f0 [snd_hda_intel] > [ 8.713646] [<ffffffffa00f98f0>] ? azx_runtime_suspend+0x40/0x40 [snd_hda_intel] > [ 8.714377] [<ffffffffa00f8800>] ? azx_remove+0x30/0x30 [snd_hda_intel] > [ 8.715111] [<ffffffffa00fc4bf>] azx_probe+0x3bf/0x7e0 [snd_hda_intel] > [ 8.715845] [<ffffffff8130b3ee>] local_pci_probe+0x3e/0x70 > [ 8.716574] [<ffffffff8130c6d1>] pci_device_probe+0x121/0x130 > [ 8.717303] [<ffffffff813bf3c7>] driver_probe_device+0x87/0x390 > [ 8.718033] [<ffffffff813bf7a3>] __driver_attach+0x93/0xa0 > [ 8.718762] [<ffffffff813bf710>] ? __device_attach+0x40/0x40 > [ 8.719481] [<ffffffff813bd303>] bus_for_each_dev+0x63/0xa0 > [ 8.720189] [<ffffffff813bee1e>] driver_attach+0x1e/0x20 > [ 8.720886] [<ffffffff813be9b8>] bus_add_driver+0x1e8/0x2a0 > [ 8.721578] [<ffffffffa016a169>] ? ftrace_define_fields_azx_get_position+0xcd/0xcd [snd_hda_intel] > [ 8.722277] [<ffffffff813bfdc4>] driver_register+0x74/0x150 > [ 8.722980] [<ffffffffa016a169>] ? ftrace_define_fields_azx_get_position+0xcd/0xcd [snd_hda_intel] > [ 8.723682] [<ffffffff8130b27b>] __pci_register_driver+0x4b/0x50 > [ 8.724368] [<ffffffffa016a187>] azx_driver_init+0x1e/0xe97 [snd_hda_intel] > [ 8.725043] [<ffffffff810002c2>] do_one_initcall+0xf2/0x1a0 > [ 8.725706] [<ffffffff8103f183>] ? set_memory_nx+0x43/0x50 > [ 8.726360] [<ffffffff810b9f3d>] load_module+0x1b9d/0x2640 > [ 8.726999] [<ffffffff810b6270>] ? store_uevent+0x40/0x40 > [ 8.727631] [<ffffffff810bab56>] SyS_finit_module+0x86/0xb0 > [ 8.728250] [<ffffffff81621782>] system_call_fastpath+0x16/0x1b > [ 8.728856] Code: ff 48 89 de 4c 89 ff 48 89 43 18 8b 85 7c ff ff ff 89 43 20 48 c7 c0 1e c5 08 a0 4c 89 45 d0 48 0f 45 c2 49 c1 e2 07 48 89 45 98 <41> 8b 44 24 18 4f 8d 34 14 89 45 8c 41 8b 86 c8 00 00 00 89 45 > [ 8.730268] RIP [<ffffffffa0086992>] snd_pcm_add_chmap_ctls+0xd2/0x160 [snd_pcm] > [ 8.730932] RSP <ffff880210599968> > [ 8.731585] CR2: 0000000000000018 > [ 8.732240] ---[ end trace 92e1db8c3a6c8fab ]--- Could you check the patch below? Thanks! Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix NULL dereference with CONFIG_SND_DYNAMIC_MINORS=n Without the dynamic minor assignment, HDMI codec may have less PCM instances than the number of pins, which eventually leads to Oops. Reported-by: Stratos Karafotis <stratosk(a)semaphore.gr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..e2cb92b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1781,6 +1781,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; + + if (pin_idx >= codec->num_pcms) + break; err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); -- 1.8.3.4

1 0

[alsa-devel] alsa-lib bug tracking
by Cornel Spinu 22 Aug '13

22 Aug '13

Hi, I'm looking for a list of bugs for alsa-lib. It seems that https://bugtrack.alsa-project.org/alsa-bug is not working. Do you have any clue where I can find this list? Thank you, Cornel

2 1

[alsa-devel] CM108 head set problem
by Nikolay Martynov 22 Aug '13

22 Aug '13

Hi. I'm experiencing a problem with CM108 based headset. The problem manifests itself in 'phone' software. When I connect the headset and run pavucontrol I can see the headset's mic in input devices and when I say something the activity bar moves - I sort of see that mic is producing some input for the OS. Then I try to make a Skype call (I've set skype to use proper input and output devices). When I make Skype call the activity bar from pavucontrol disappears and mic stops working - the other side of the call doesn't hear me. Similar thing happens with Google hangout. The only difference is that with Skype it happens immediately and with Google hangout it looks like it only happens when I do something 'heavy'. The thing that often triggers the problem with Google hangout is switching desktops (I have compiz running). I'm not entirely sure that problem is alsa related but I have this in dmesg: [72856.844289] usb 3-1: USB disconnect, device number 10 [72858.632109] usb 3-1: new full-speed USB device number 11 using uhci_hcd [72858.792090] usb 3-1: New USB device found, idVendor=045e, idProduct=070f [72858.792098] usb 3-1: New USB device strings: Mfr=0, Product=1, SerialNumber=0 [72858.792103] usb 3-1: Product: Microsoft LifeChat LX-3000 [72858.875205] input: Microsoft LifeChat LX-3000 as /devices/pci0000:00/0000:00:1d.1/usb3/3-1/3-1:1.3/input/input19 [72858.875399] hid-generic 0003:045E:070F.000B: input,hidraw0: USB HID v1.00 Device [Microsoft LifeChat LX-3000 ] on usb-0000:00:1d.1-1/input3 [72876.015919] retire_capture_urb: 22 callbacks suppressed [72876.411354] delay: estimated 133, actual 1 [72876.496315] cannot submit urb (err = -18) [72876.496330] cannot submit urb (err = -18) [72876.496341] cannot submit urb (err = -18) [72876.496349] cannot submit urb (err = -18) [72876.496357] cannot submit urb (err = -18) [72876.496366] cannot submit urb (err = -18) [72876.496375] cannot submit urb (err = -18) [72876.496383] cannot submit urb (err = -18) [72876.499076] delay: estimated 222, actual 1 [72876.587212] delay: estimated 90, actual 1 [72876.627072] delay: estimated 266, actual 1 [72876.675644] delay: estimated 353, actual 0 [72876.747102] delay: estimated 354, actual 1 [72876.851132] delay: estimated 353, actual 1 [72877.059072] delay: estimated 90, actual 1 [72877.171110] delay: estimated 353, actual 1 [72877.219082] delay: estimated 90, actual 1 [72877.299493] delay: estimated 310, actual 1 [72877.547100] delay: estimated 354, actual 1 It looks like once I get error like message 'cannot submit urb (err = -18)' mic stops working. I would really appreciate any help solving this problem and I would be happy to do any testing required or provide any additional information. I'm using Linux Mint 15 (Ubuntu 13.04) with '3.10.5' kernel from Ubuntu mainline kernel builds. The very same problem also happens with 'standard' kernel that comes with Ubuntu. I have a dell inspiron 6400 laptop with core2duo CPU and ati video. Thanks! -- Martynov Nikolay. Email: mar.kolya(a)gmail.com

1 0

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 21 Aug '13

21 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 200 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/rt5640.h | 11 +++ 2 files changed, 157 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..218aae2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_R_MUTE_SFT, 1, 1); + /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +926,103 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(30000, 35000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2 , + RT5640_PWR_FV1 | RT5640_PWR_FV2); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK, + RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN); + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + + regmap_write(rt5640->regmap, RT5640_PR_BASE + + RT5640_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK, + RT5640_HP_CP_PD | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val = 0, pow; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + if (!(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (!(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(35000, 40000); + break; + + case SND_SOC_DAPM_PRE_PMD: + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_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: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1212,27 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1381,9 +1473,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, @@ -1396,13 +1490,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,8 +1508,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, - {"HPOL", NULL, "HP L Amp"}, - {"HPOR", NULL, "HP R Amp"}, + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, {"MONOP", NULL, "Mono MIX"}, @@ -1792,17 +1890,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +1940,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..4d2cfe7 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) -- 1.8.1.1.439.g50a6b54

4 4

[alsa-devel] [PATCH v8 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 21 Aug '13

21 Aug '13

Changelog: v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1281 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1740 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 15

[alsa-devel] [PATCH v9 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 20 Aug '13

20 Aug '13

Changelog: v8->v9: * Use bool instead of atomic_t. * Use clk_prepare_enable() instead. * Dropped dumpregs(). * Dropped unnecessary clk_round_rate(). * Dropped unused clock source enum. * Revised dev_err() message. v7->v8: * Revised DT binding doc. * Revised dev_err() error msg. * Check return value of clk_set_rate(). v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 54 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1236 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 191 +++ sound/soc/fsl/imx-spdif.c | 134 +++ 7 files changed, 1662 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

4 7

[alsa-devel] [PATCH v7 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 20 Aug '13

20 Aug '13

Changelog: v6->v7: * Removed extra comma in array. * Revised some comments. * Added some dev_err(). * Check the return value of soft reset. * Use bitrev8() instead of extra reverse_bits(). * Use cache_bypass as default for regmap. v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 56 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1277 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1738 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

6 21

[alsa-devel] [PATCH 0/6] simplify platform_get_resource_byname/devm_ioremap_resource
by Julia Lawall 20 Aug '13

20 Aug '13

devm_ioremap_resource often uses the result of a call to platform_get_resource_byname as its last argument. devm_ioremap_resource does appropriate error handling on this argument, so error handling can be removed from the call to platform_get_resource_byname. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l,f; expression list es; @@ ( res = f(...); - if (res == NULL) { ... $goto l;\|return ret;$ } e = devm_ioremap_resource(e1, res); | - res = f(es); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = f(es); e = devm_ioremap_resource(e1, res); ) // </smpl> In practice, f is always platform_get_resource_byname (or platform_get_resource, which was handled by a previous patch series). And the call to platform_get_resource_byname always immediately precedes the call to devm_ioremap_resource.

3 3

[alsa-devel] [PATCH] ASoC: fsl: fsl_ssi: Fix the order of resources removal
by Fabio Estevam 20 Aug '13

20 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> In fsl_ssi_remove() we need to remove the resources in the opposite order that they were acquired in probe. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0c072ff..c8aee4e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -928,14 +928,14 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (!ssi_private->new_binding) platform_device_unregister(ssi_private->pdev); - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); - clk_disable_unprepare(ssi_private->clk); - } snd_soc_unregister_component(&pdev->dev); - device_remove_file(&pdev->dev, &ssi_private->dev_attr); - irq_dispose_mapping(ssi_private->irq); dev_set_drvdata(&pdev->dev, NULL); + device_remove_file(&pdev->dev, &ssi_private->dev_attr); + if (ssi_private->ssi_on_imx) + clk_disable_unprepare(ssi_private->clk); + irq_dispose_mapping(ssi_private->irq); return 0; } -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: samsung: Rename DMA platform registration functions
by Mark Brown 20 Aug '13

20 Aug '13

From: Mark Brown <broonie(a)linaro.org> The current naming with a simple asoc_ prefix is too generic for use in multiplatform kernels. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/dma.c | 8 ++++---- sound/soc/samsung/dma.h | 4 ++-- sound/soc/samsung/i2s.c | 6 +++--- 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index c732df9..2acf987 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -457,7 +457,7 @@ static int s3c_ac97_probe(struct platform_device *pdev) if (ret) goto err5; - ret = asoc_dma_platform_register(&pdev->dev); + ret = samsung_asoc_dma_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "failed to get register DMA: %d\n", ret); goto err6; @@ -480,7 +480,7 @@ static int s3c_ac97_remove(struct platform_device *pdev) { struct resource *irq_res; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index d54df85..9338d11 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -444,17 +444,17 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; -int asoc_dma_platform_register(struct device *dev) +int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_register); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_register); -void asoc_dma_platform_unregister(struct device *dev) +void samsung_asoc_dma_platform_unregister(struct device *dev) { snd_soc_unregister_platform(dev); } -EXPORT_SYMBOL_GPL(asoc_dma_platform_unregister); +EXPORT_SYMBOL_GPL(samsung_asoc_dma_platform_unregister); MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); MODULE_DESCRIPTION("Samsung ASoC DMA Driver"); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 189a7a6..0e86315 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,7 +22,7 @@ struct s3c_dma_params { char *ch_name; }; -int asoc_dma_platform_register(struct device *dev); -void asoc_dma_platform_unregister(struct device *dev); +int samsung_asoc_dma_platform_register(struct device *dev); +void samsung_asoc_dma_platform_unregister(struct device *dev); #endif diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ce487c2..b302f3b 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1146,7 +1146,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) snd_soc_register_component(&sec_dai->pdev->dev, &samsung_i2s_component, &sec_dai->i2s_dai_drv, 1); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1263,7 +1263,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); - asoc_dma_platform_register(&pdev->dev); + samsung_asoc_dma_platform_register(&pdev->dev); return 0; err: @@ -1293,7 +1293,7 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->pri_dai = NULL; i2s->sec_dai = NULL; - asoc_dma_platform_unregister(&pdev->dev); + samsung_asoc_dma_platform_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); return 0; -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ASoC: samsung: Check to see if we managed to allocate a channel
by Mark Brown 20 Aug '13

20 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/dma.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 6e2b2b4..d54df85 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -183,6 +183,10 @@ static int dma_hw_params(struct snd_pcm_substream *substream, prtd->params->ch = prtd->params->ops->request( prtd->params->channel, &req, rtd->cpu_dai->dev, prtd->params->ch_name); + if (!prtd->params->ch) { + pr_err("Failed to allocate DMA channel\n"); + return -ENXIO; + } prtd->params->ops->config(prtd->params->ch, &config); } -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH RFC 0/9] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 19 Aug '13

19 Aug '13

Hi, This patch series attempts to fix a long starting problem with the concurrent usage of playback and capture on implicit feedback devices. As Clemens found out, when implicit feedback is used, playback and capture cannot be used in the same time - unless started at the same time (e.g., with jack). While a substream is active, attempting to start a substream in the opposite direction will fail and also (on some devices?) cause the currently active stream to stop. There are multiple issues that contribute to this: 1. URBs are unconditionally deactivated in hw_free. On implicit feedback devices, the sync_endpoint (= data_endpoint of the capture) will be deactivated here. Fixed in patch #4. 2. Changing the alternate settings of an interface might break currently running streams. This happens on my device, but I suspect other devices too. Fixed in patch #7. 3. The endpoint use_count has a subtle issue (see patch #9). This causes jack to work fine, while using two separate programs (most probably) fails. In short: set_format is called from hw_params, but use_count is incremented only later (prepare or trigger). 4. Concurrent usage should be allowed but only if the hw_params match up (patch #8) One problem with this series is that I query the device for the altsettting. This might not be reliable for all devices, so do I plan to remove this. Of course, I've only tested this on my device, but it seems to work good and I couldn't break it so far, so hopefully I got this right. Can someone give it a try on other devices (also on non-implicit-feedback)? Cheers, Eldad Eldad Zack (9): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: correct ep use_count semantics (add set_param flag) sound/usb/card.h | 1 + sound/usb/endpoint.c | 65 +++++++++++++++------ sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 155 +++++++++++++++++++++++++++++++++++++++------------ 4 files changed, 176 insertions(+), 56 deletions(-) -- 1.8.1.5

2 11

[alsa-devel] [PATCH 0/3] HDSPM: Fix LTC ioctl
by Adrian Knoth 19 Aug '13

19 Aug '13

Hi! This series fixes the broken LTC ioctl to query the TCO. Apparently, nobody ever attempted to use the code, otherwise, this rather obvious problem would have been spotted way earlier. This is maybe because not too many people happen to have a TV studio around to make use of this feature. ;) I'll also send a patch to fix alsa-lib. Cheers Adrian Knoth (3): ALSA: hdspm - Fix default value in SNDRV_HDSPM_IOCTL_GET_LTC ALSA: hdspm - Use enums in hdspm_tco_ltc_frames() ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTC include/uapi/sound/hdspm.h | 2 +- sound/pci/rme9652/hdspm.c | 12 ++++++------ 2 files changed, 7 insertions(+), 7 deletions(-) -- 1.8.3.2

2 4

[alsa-devel] [PATCH 0/1] alsa-lib: update hdspm.h
by Adrian Knoth 19 Aug '13

19 Aug '13

Hi! Here's the patch for alsa-lib to fix the broken LTC ioctl. Cheers Adrian Knoth (1): hdspm.h: Update LTC ioctl to use struct hdspm_ltc include/sound/hdspm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) -- 1.8.3.2

1 1

Re: [alsa-devel] [PATCH v11 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 19 Aug '13

19 Aug '13

On Mon, Aug 19, 2013 at 05:13:57PM +0200, Marc Kleine-Budde wrote: > On 08/19/2013 05:13 PM, Marc Kleine-Budde wrote: > > On 08/19/2013 05:05 PM, Markus Pargmann wrote: > >> Michael Grzeschik (2): > >> ASoC: fsl-ssi: imx-pcm-fiq bugfix > >> ASoC: fsl: disable ssi irq for imx > >> > >> Steffen Trumtrar (1): > >> ASoC: fsl-ssi: add SSIEN errata work around > > > > You should add you S-o-b to patches from other develops. > developers yes, I forgot that, thanks Markus -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

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Re: [alsa-devel] [PATCH 3.11-rc5 1/2] ALSA: pcm: Add snd_printd_ratelimit()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Fri, 16 Aug 2013 11:18:59 -0600, Tim Gardner wrote: > > Direct calls to printk_limit() will emit log noise even when CONFIG_SND_DEBUG is not > defined. Add a wrapper macro around printk_limit() that is conditionally defined by > CONFIG_SND_DEBUG. > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Cc: Lars-Peter Clausen <lars(a)metafoo.de> > Cc: Yacine Belkadi <yacine.belkadi.1(a)gmail.com> > Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> > --- > > checkpatch is moaning about the use of printk_ratelimit(), but > printk_ratelimited() seems like overkill for a debug feature. > > WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit > #38: FILE: include/sound/core.h:382: > +#define snd_printd_ratelimit() printk_ratelimit() Thanks, applied both patches now. But it's no real bug, so I postpone this for 3.12 inclusion. Takashi > > include/sound/core.h | 8 ++++++++ > 1 file changed, 8 insertions(+) > > diff --git a/include/sound/core.h b/include/sound/core.h > index c586617..2a14f1f 100644 > --- a/include/sound/core.h > +++ b/include/sound/core.h > @@ -27,6 +27,7 @@ > #include <linux/rwsem.h> /* struct rw_semaphore */ > #include <linux/pm.h> /* pm_message_t */ > #include <linux/stringify.h> > +#include <linux/printk.h> > > /* number of supported soundcards */ > #ifdef CONFIG_SND_DYNAMIC_MINORS > @@ -376,6 +377,11 @@ void __snd_printk(unsigned int level, const char *file, int line, > #define snd_BUG() WARN(1, "BUG?\n") > > /** > + * Suppress high rates of output when CONFIG_SND_DEBUG is enabled. > + */ > +#define snd_printd_ratelimit() printk_ratelimit() > + > +/** > * snd_BUG_ON - debugging check macro > * @cond: condition to evaluate > * > @@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {} > unlikely(__ret_warn_on); \ > }) > > +static inline bool snd_printd_ratelimit(void) { return false; } > + > #endif /* CONFIG_SND_DEBUG */ > > #ifdef CONFIG_SND_DEBUG_VERBOSE > -- > 1.7.9.5 >

1 0

Re: [alsa-devel] [PATCH v1 45/49] sound: usb: caiaq: prepare for enabling irq in complete()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Sun, 18 Aug 2013 00:25:10 +0800, Ming Lei wrote: > > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> Acked-by: Takashi Iwai <tiwai(a)suse.e> thanks, Takashi > Cc: alsa-devel(a)alsa-project.org > Acked-by: Daniel Mack <zonque(a)gmail.com> > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > sound/usb/caiaq/audio.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c > index 7103b09..e5675ab 100644 > --- a/sound/usb/caiaq/audio.c > +++ b/sound/usb/caiaq/audio.c > @@ -672,10 +672,11 @@ static void read_completed(struct urb *urb) > offset += len; > > if (len > 0) { > - spin_lock(&cdev->spinlock); > + unsigned long flags; > + spin_lock_irqsave(&cdev->spinlock, flags); > fill_out_urb(cdev, out, &out->iso_frame_desc[outframe]); > read_in_urb(cdev, urb, &urb->iso_frame_desc[frame]); > - spin_unlock(&cdev->spinlock); > + spin_unlock_irqrestore(&cdev->spinlock, flags); > check_for_elapsed_periods(cdev, cdev->sub_playback); > check_for_elapsed_periods(cdev, cdev->sub_capture); > send_it = 1; > -- > 1.7.9.5 >

1 0

Re: [alsa-devel] [PATCH v1 44/49] sound: usb: midi: prepare for enabling irq in complete()
by Takashi Iwai 19 Aug '13

19 Aug '13

At Sun, 18 Aug 2013 00:25:09 +0800, Ming Lei wrote: > > Complete() will be run with interrupt enabled, so change to > spin_lock_irqsave(). > > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> Acked-by: Takashi Iwai <tiwai(a)suse.de> thanks, Takashi > Cc: Clemens Ladisch <clemens(a)ladisch.de> > Cc: alsa-devel(a)alsa-project.org > Signed-off-by: Ming Lei <ming.lei(a)canonical.com> > --- > sound/usb/midi.c | 5 +++-- > 1 file changed, 3 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/midi.c b/sound/usb/midi.c > index b901f46..86af276 100644 > --- a/sound/usb/midi.c > +++ b/sound/usb/midi.c > @@ -279,15 +279,16 @@ static void snd_usbmidi_out_urb_complete(struct urb* urb) > struct out_urb_context *context = urb->context; > struct snd_usb_midi_out_endpoint* ep = context->ep; > unsigned int urb_index; > + unsigned long flags; > > - spin_lock(&ep->buffer_lock); > + spin_lock_irqsave(&ep->buffer_lock, flags); > urb_index = context - ep->urbs; > ep->active_urbs &= ~(1 << urb_index); > if (unlikely(ep->drain_urbs)) { > ep->drain_urbs &= ~(1 << urb_index); > wake_up(&ep->drain_wait); > } > - spin_unlock(&ep->buffer_lock); > + spin_unlock_irqrestore(&ep->buffer_lock, flags); > if (urb->status < 0) { > int err = snd_usbmidi_urb_error(urb->status); > if (err < 0) { > -- > 1.7.9.5 >

1 0

[alsa-devel] [PATCH] ALSA: hda - Try to allow haswell HDMI audio even without powerwell
by David Henningsson 19 Aug '13

19 Aug '13

If compiled without CONFIG_SND_HDA_I915, the audio driver cannot request power well. However, if the power well is on for other reasons, maybe audio can still work. Therefore, do not skip the card completely if compiled without CONFIG_SND_HDA_I915. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) I'm going to apply this on my local DKMS tree, it's important for our backports as we combine different versions of the kernel with backports for haswell etc. I'm not sure if this is beneficial to have on 3.11+ kernels as well, but I guess it could be. At least it shouldn't harm. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f9e406..c6c9829 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3855,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip) /* Request power well for Haswell HDA controller and codec */ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { +#ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(); if (err < 0) { snd_printk(KERN_ERR SFX "Error request power-well from i915\n"); goto out_free; } +#endif hda_display_power(true); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Limit internal mic boost for a few more Thinkpad machines
by David Henningsson 19 Aug '13

19 Aug '13

The higher mic boosts (on internal mic) are so noisy they're unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213820 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d67a48c..00d00b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,9 +3704,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: ac97: Provide stub DAPM integration
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ac97.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ec73518..8d9ba4ba 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -23,6 +23,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget ac97_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ac97_routes[] = { + { "AC97 Capture", NULL, "RX" }, + { "TX", NULL, "AC97 Playback" }, +}; + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -117,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, + + .dapm_widgets = ac97_widgets, + .num_dapm_widgets = ARRAY_SIZE(ac97_widgets), + .dapm_routes = ac97_routes, + .num_dapm_routes = ARRAY_SIZE(ac97_routes), }; static int ac97_probe(struct platform_device *pdev) -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ASoC: hdmi: Provide stub DAPM integration
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/hdmi.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index f0986b9..68342b1 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -23,11 +23,20 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver hdmi_codec; +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; static struct snd_soc_dai_driver hdmi_codec_dai = { .name = "hdmi-hifi", .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | @@ -38,6 +47,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { SNDRV_PCM_FMTBIT_S24_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | @@ -50,6 +60,13 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { }; +static struct snd_soc_codec_driver hdmi_codec = { + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + static int hdmi_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &hdmi_codec, -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/2] ASoC: bt-sco: Add generic compatible string
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Provide a common compatible string for device trees to list as a fallback for simplicity. We don't currently have a binding document but let's not fix that right now... Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/bt-sco.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index a081d9f..5c040ce 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -50,6 +50,9 @@ static struct platform_device_id bt_sco_driver_ids[] = { { .name = "dfbmcs320", }, + { + .name = "bt-sco", + }, {}, }; MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); -- 1.8.4.rc3

1 1

[alsa-devel] [PATCH] ASoC: spdif: Add stub DAPM widgets for Rx
by Mark Brown 19 Aug '13

19 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure that the driver continues to work with mandatory DAPM. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/spdif_receiver.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index 26d3474..d5463c0 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -23,13 +23,26 @@ #include <sound/initval.h> #include <linux/of.h> +static const struct snd_soc_dapm_widget dir_widgets[] = { + SND_SOC_DAPM_INPUT("spdif-in"), +}; + +static const const struct snd_soc_dapm_route dir_routes[] = { + { "Capture", NULL, "spdif-in" }, +}; + #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dir; +static struct snd_soc_codec_driver soc_codec_spdif_dir = { + .dapm_widgets = dir_widgets, + .num_dapm_widgets = ARRAY_SIZE(dir_widgets), + .dapm_routes = dir_routes, + .num_dapm_routes = ARRAY_SIZE(dir_routes), +}; static struct snd_soc_dai_driver dir_stub_dai = { .name = "dir-hifi", -- 1.8.4.rc3

1 0

[alsa-devel] [PATCH] ALSA: hda - Limit internal mic boost for another Thinkpad machine
by David Henningsson 19 Aug '13

19 Aug '13

The higher mic boosts (on internal mic) are so noisy they're unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213820 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d67a48c..ef09fbc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,6 +3704,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 1.7.9.5

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Re: [alsa-devel] [PATCH v6 1/2] ASoC: fsl: Add S/PDIF CPU DAI driver
by Nicolin Chen 19 Aug '13

19 Aug '13

On Mon, Aug 19, 2013 at 02:31:47PM +0800, Bhushan Bharat-R65777 wrote: > > > > We here suppose the reset bit would be cleared -- "The software > > > > reset will last > > > > 8 cycles." from RM, so if this happened to be a failure, the whole > > > > IP module won't be normally working as well. > > > > > > Also add a comment describing this against why cycle = 1000 is selected. > > > > If it is done in 8 cycles, 1000-cycle will be surely a safe value for it. > > As long as it finished in 8 cycles, it would quit anyway. Why against? > > I am not against, I am saying why it was not 200 or 50 or 20 etc. I am saying that write a comment saying this much is sufficient as per specification and so keep 1000/etc as preservative. I did't mean that. The 'against' is from "Also add a comment describing this 'against' why cycle = 1000 is selected." Well, if you insist this extra comment for easy-understand, I'll add them Thank you. Nicolin Chen

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Re: [alsa-devel] [PATCH v6 1/2] ASoC: fsl: Add S/PDIF CPU DAI driver
by Nicolin Chen 19 Aug '13

19 Aug '13

Hi Bhushan, I'll revise some as you suggest. Just a few replies here. On Mon, Aug 19, 2013 at 12:38:11PM +0800, Bhushan Bharat-R65777 wrote: > > We here suppose the reset bit would be cleared -- "The software reset will last > > 8 cycles." from RM, so if this happened to be a failure, the whole IP module > > won't be normally working as well. > > Also add a comment describing this against why cycle = 1000 is selected. If it is done in 8 cycles, 1000-cycle will be surely a safe value for it. As long as it finished in 8 cycles, it would quit anyway. Why against? > > > > +static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg) > > > > +{ > > > > + /* Sync all registers after reset */ > > > > > > Where us sync :) ? > > > > The "return true" would do that. For volatile registers, if no "return true" > > here, the whole regmap would use the value in cache, while for some bits > > we need to trace its true value from the physical registers not from cache. > > Where will be device registers cached? Do not we program them to be non-cacheable in core? regmap has a regcache for all the mapped registers. Set the regsiters as volatile will allow the driver to sync the regcache with physical memory each time when using regmap_read/write/update_bits(). But I think I can try to use the regcache_bypass instead. Thank you, Nicolin Chen

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Re: [alsa-devel] [PATCH v6 1/2] ASoC: fsl: Add S/PDIF CPU DAI driver
by Nicolin Chen 19 Aug '13

19 Aug '13

Hi Bhushan, Thank you for the comments :) I'll fix some in v7. Here is my some replies to you. On Sat, Aug 17, 2013 at 02:24:19AM +0800, Bhushan Bharat-R65777 wrote: > > This patch add S/PDIF controller driver for Freescale SoC. > > Please give some more description of the driver? I've referred some ASoC drivers, all of them seem to be brief as mine. So I'm not sure what else information I should provide here. It's already kinda okay to me. > > +struct spdif_mixer_control { > > + /* buffer ptrs for writer */ > > + u32 upos; > > + u32 qpos; > > They does not look like pointer? They are more like offsets to get the correspond pointer. But I'll change the confusing comments. > > +/* U/Q Channel receive register full */ > > +static void spdif_irq_uqrx_full(struct fsl_spdif_priv *spdif_priv, char name) > > +{ > > + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; > > + struct regmap *regmap = spdif_priv->regmap; > > + struct platform_device *pdev = spdif_priv->pdev; > > + u32 *pos, size, val, reg; > > + > > + switch (name) { > > + case 'U': > > + pos = &ctrl->upos; > > + size = SPDIF_UBITS_SIZE; > > + reg = REG_SPDIF_SRU; > > + break; > > + case 'Q': > > + pos = &ctrl->qpos; > > + size = SPDIF_QSUB_SIZE; > > + reg = REG_SPDIF_SRQ; > > + break; > > + default: > > + return; > > Should return error. IMHO, this should be fine. It's a void type function and being used in the isr(). The params 'name' is totally controlled by driver itself, so basically we don't need to worry about the default path. > > + if (*pos >= size * 2) { > > + *pos = 0; > > + } else if (unlikely((*pos % size) + 3 > size)) { > > + dev_err(&pdev->dev, "User bit receivce buffer overflow\n"); > > + return; > > Should return error. Ditto, it's being used in isr(), we don't need to detect the return value, just use dev_err() to warn users and let the driver clear the irq. > > +/* U/Q Channel framing error */ > > +static void spdif_irq_uq_err(struct fsl_spdif_priv *spdif_priv) > > +{ > > + struct spdif_mixer_control *ctrl = &spdif_priv->fsl_spdif_control; > > + struct regmap *regmap = spdif_priv->regmap; > > + struct platform_device *pdev = spdif_priv->pdev; > > + u32 val; > > + > > + dev_dbg(&pdev->dev, "isr: U/Q Channel framing error\n"); > > + > > + /* read U/Q data and do buffer reset */ > > + regmap_read(regmap, REG_SPDIF_SRU, &val); > > + regmap_read(regmap, REG_SPDIF_SRQ, &val); > > Above prints says read u/q data and buffer reset, what is buffer reset? Is that read on clear? That's the behavior needed by IP, according to the reference manual: "U Channel receive register full, can't be cleared with reg. IntClear. To clear it, read from U Rx reg." and "Q Channel receive register full, can't be cleared with reg. IntClear. To clear it, read from Q Rx reg." > > +static void spdif_softreset(struct fsl_spdif_priv *spdif_priv) > > +{ > > + struct regmap *regmap = spdif_priv->regmap; > > + u32 val, cycle = 1000; > > + > > + regmap_write(regmap, REG_SPDIF_SCR, SCR_SOFT_RESET); > > + regcache_sync(regmap); > > + > > + /* RESET bit would be cleared after finishing its reset procedure */ > > + do { > > + regmap_read(regmap, REG_SPDIF_SCR, &val); > > + } while ((val & SCR_SOFT_RESET) && cycle--); > > What if reset is not cleared and timeout happen? We here suppose the reset bit would be cleared -- "The software reset will last 8 cycles." from RM, so if this happened to be a failure, the whole IP module won't be normally working as well. Well, but I don't mind to put here an extra failed return to make it clear. > > +static u8 reverse_bits(u8 input) > > +{ > > + u8 tmp = input; > > + > > + tmp = ((tmp & 0b10101010) >> 1) | ((tmp << 1) & 0b10101010); > > + tmp = ((tmp & 0b11001100) >> 2) | ((tmp << 2) & 0b11001100); > > + tmp = ((tmp & 0b11110000) >> 4) | ((tmp << 4) & 0b11110000); > > What is this logic, can the hardcoding be removed and some description on above calculation? This was provided by Philipp Zabel in his review at patch v3. It's pretty clear to me that it just reverses the bits for u8. I don't think this logic has any problem and the mask here doesn't look like any hardcode to me. > > +static bool fsl_spdif_volatile_reg(struct device *dev, unsigned int reg) > > +{ > > + /* Sync all registers after reset */ > > Where us sync :) ? The "return true" would do that. For volatile registers, if no "return true" here, the whole regmap would use the value in cache, while for some bits we need to trace its true value from the physical registers not from cache. Best regards, Nicolin Chen

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[alsa-devel] [PATCH 0/4] clk: Samsung: audss: Add support for Exynos5420
by Padmavathi Venna 18 Aug '13

18 Aug '13

This patch set adds support for audio subsystem clks on Exynos5420. Exynos5420 audio subsystem has a gate bit for ADMA controller and the some of parent clks for mout_i2s are also different from Exynos5250. So this patch adds provision for supporting both the platforms by passing the parent clk names through device tree. Andrew Bresticker (3): clk: exynos-audss: add support for Exynos 5420 clk: exynos-audss: allow input clocks to be specified in device tree ARM: dts: exynos5420: add audio clock controller Padmavathi Venna (1): ARM: dts: Correct the /include entry on exynos5420 dtsi file .../devicetree/bindings/clock/clk-exynos-audss.txt | 38 +++++++++++++++++-- arch/arm/boot/dts/exynos5420.dtsi | 13 ++++++- drivers/clk/samsung/clk-exynos-audss.c | 36 ++++++++++++++++-- include/dt-bindings/clk/exynos-audss-clk.h | 3 +- 4 files changed, 80 insertions(+), 10 deletions(-) -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: wm8782: Add DAPM support
by Mark Brown 18 Aug '13

18 Aug '13

From: Mark Brown <broonie(a)linaro.org> In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8782.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f1fdbf6..8092495 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -26,6 +26,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route wm8782_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + static struct snd_soc_dai_driver wm8782_dai = { .name = "wm8782", .capture = { @@ -40,7 +50,12 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8782; +static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { + .dapm_widgets = wm8782_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), + .dapm_routes = wm8782_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes), +}; static int wm8782_probe(struct platform_device *pdev) { -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: wm8727: Add DAPM support
by Mark Brown 18 Aug '13

18 Aug '13

From: Mark Brown <broonie(a)linaro.org> In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8727.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 462f5e4..7b1a6d5 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -23,6 +23,16 @@ #include <sound/initval.h> #include <sound/soc.h> +static const struct snd_soc_dapm_widget wm8727_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route wm8727_dapm_routes[] = { + { "VOUTL", NULL, "Playback" }, + { "VOUTR", NULL, "Playback" }, +}; + /* * Note this is a simple chip with no configuration interface, sample rate is * determined automatically by examining the Master clock and Bit clock ratios @@ -43,7 +53,12 @@ static struct snd_soc_dai_driver wm8727_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8727; +static struct snd_soc_codec_driver soc_codec_dev_wm8727 = { + .dapm_widgets = wm8727_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets), + .dapm_routes = wm8727_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8727_dapm_routes), +}; static int wm8727_probe(struct platform_device *pdev) { -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: wl1273: Add stub DAPM support
by Mark Brown 18 Aug '13

18 Aug '13

From: Mark Brown <broonie(a)linaro.org> In order to ensure that the device continues to work with DAPM support being mandatory provide stub DAPM widgets and routes. Note that the public information on the device appears to make no mention of the FM support the driver appears to have. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wl1273.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 54cd3da..b7ab2ef 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -290,6 +290,18 @@ static const struct snd_kcontrol_new wl1273_controls[] = { snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), }; +static const struct snd_soc_dapm_widget wl1273_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route wl1273_dapm_routes[] = { + { "Capture", NULL, "RX" }, + + { "TX", NULL, "Playback" }, +}; + static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -483,6 +495,11 @@ static int wl1273_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { .probe = wl1273_probe, .remove = wl1273_remove, + + .dapm_widgets = wl1273_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets), + .dapm_routes = wl1273_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wl1273_dapm_routes), }; static int wl1273_platform_probe(struct platform_device *pdev) -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/1] ASoC: wm8960: Fix PLL register writes
by Mike Dyer 18 Aug '13

18 Aug '13

Bit 9 of PLL2,3 and 4 is reserved as '0'. The 24bit fractional part should be split across each register in 8bit chunks. Signed-off-by: Mike Dyer <mike.dyer(a)md-soft.co.uk> --- sound/soc/codecs/wm8960.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd..5e5af89 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, if (pll_div.k) { reg |= 0x20; - snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff); } snd_soc_write(codec, WM8960_PLL1, reg); -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: fsl: Drop SND_SOC_FSL_UTILS from i.mx machine code
by Fabio Estevam 18 Aug '13

18 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the i.mx case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index e15f771..3a4808d 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_IMX_WM8962 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a wm8962 codec. @@ -187,7 +186,6 @@ config SND_SOC_IMX_SGTL5000 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. -- 1.8.1.2

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[alsa-devel] [PATCH v4 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 17 Aug '13

17 Aug '13

Changelog: v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 100 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1350 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 223 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1854 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH v6 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 16 Aug '13

16 Aug '13

Changelog: v5->v6: * Sorted out rxtx clk source in DT binding. * Use devm_xxxx() functions. * Use platform_get_resource() instead. v4->v5: * Dropped rx/tx-clksrc-names DT bindings. * Use standard clock binding instead to pass the clock source list. * Update the compatible list by using "imx35", the first SoC that has spdif. v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 56 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1272 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 224 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1733 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH 1/2] ALSA: hda - Fix internal mic boost on three Thinkpad machines
by David Henningsson 16 Aug '13

16 Aug '13

The internal mic boost is so noisy on boosts 2 and 3 so they are unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213055 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ae9504..6ec96f9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3704,6 +3704,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 1.7.9.5

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[alsa-devel] [git pull] [PATCH 0/7] add support for many Roland and Yamaha devices
by Clemens Ladisch 16 Aug '13

16 Aug '13

The following changes since commit f722406faae2d073cc1d01063d1123c35425939e: Linux 3.10-rc1 (2013-05-11 17:14:08 -0700) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git full-roland-support for you to fetch changes up to b7f33917bcd993ff81f3f80b9dc1890fb7410c6d: ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE (2013-06-27 21:59:50 +0200) ---------------------------------------------------------------- Clemens Ladisch (7): ALSA: usb-audio: store protocol version in struct audioformat ALSA: usb-audio: detect implicit feedback on Roland devices ALSA: usb-audio: add support for many Roland/Yamaha devices ALSA: usb-audio: add MIDI port names for some Roland devices ALSA: usb-audio: remove superfluous Roland quirks ALSA: usb-audio: claim autodetected PCM interfaces all at once ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE sound/usb/card.h | 1 + sound/usb/clock.c | 4 +- sound/usb/format.c | 34 +--- sound/usb/format.h | 2 +- sound/usb/midi.c | 74 +++++++ sound/usb/pcm.c | 45 ++++- sound/usb/quirks-table.h | 497 +++++++++-------------------------------------- sound/usb/quirks.c | 209 ++++++++++++++++++++ sound/usb/stream.c | 18 +- sound/usb/usbaudio.h | 2 + 10 files changed, 445 insertions(+), 441 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Remove direct use of internal I/O functions
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> Use the core to do I/O rather than directly calling the driver operations in order to support further refactoring. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/tlv320aic26.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b192cd4..a4f9360 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -174,9 +174,9 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg); reg = dval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); @@ -185,13 +185,13 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, reg |= 0x0800; if (fsref == 48000) reg |= 0x2000; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Audio Control 1 (FSref divisor) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); reg &= ~0x0fff; reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); return 0; } @@ -212,7 +212,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) reg |= 0x8080; else reg &= ~0x8080; - aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg); + snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); return 0; } @@ -348,7 +348,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -368,20 +368,20 @@ static int aic26_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ - aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00); + snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); /* Power up CODEC */ - aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0); + snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3); + reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3); reg &= ~0xf800; reg |= 0x0800; /* set master mode */ - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Fill register cache */ for (i = 0; i < codec->driver->reg_cache_size; i++) - aic26_reg_read(codec, i); + snd_soc_read(codec, i); /* Register the sysfs files for debugging */ /* Create SysFS files */ -- 1.8.4.rc2

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[alsa-devel] [PATCH] ASoC: imx: Add MODULE_LICENSE to DMA drivers
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> Reported-by: Ben Hutchings <ben(a)decadent.org.uk> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/imx-pcm-dma.c | 2 ++ sound/soc/fsl/imx-pcm-fiq.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index f323ce0..aa6de5c 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -73,3 +73,5 @@ void imx_pcm_dma_exit(struct platform_device *pdev) snd_dmaengine_pcm_unregister(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_dma_exit); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 3b2ba99..34043c5 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -408,3 +408,5 @@ void imx_pcm_fiq_exit(struct platform_device *pdev) snd_soc_unregister_platform(&pdev->dev); } EXPORT_SYMBOL_GPL(imx_pcm_fiq_exit); + +MODULE_LICENSE("GPL"); -- 1.8.4.rc2

1 0

[alsa-devel] [PATCH] ASoC: si476x: Add DAPM support
by Mark Brown 16 Aug '13

16 Aug '13

From: Mark Brown <broonie(a)linaro.org> This ensures the driver continues to work with DAPM mandatory and makes it easier to connect the device up to other components in the subsystem. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/si476x.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 73e205c..38f3b10 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -102,6 +102,16 @@ static int si476x_codec_write(struct snd_soc_codec *codec, return err; } +static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +}; + +static const struct snd_soc_dapm_route si476x_dapm_routes[] = { + { "Capture", NULL, "LOUT" }, + { "Capture", NULL, "ROUT" }, +}; + static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -260,6 +270,10 @@ static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, .read = si476x_codec_read, .write = si476x_codec_write, + .dapm_widgets = si476x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), + .dapm_routes = si476x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes), }; static int si476x_platform_probe(struct platform_device *pdev) -- 1.8.4.rc2

1 0

[alsa-devel] [PATCH] ASoC: imx-audmux: Move definitions to dt-bindings
by Markus Pargmann 16 Aug '13

16 Aug '13

Move imx-audmux macro definitions to include/dt-bindings, so they can be used for devicetree. Signed-off-by: Markus Pargmann <mpa(a)pengutronix.de> --- include/dt-bindings/sound/fsl-imx-audmux.h | 56 ++++++++++++++++++++++++++++++ sound/soc/fsl/imx-audmux.h | 52 +-------------------------- 2 files changed, 57 insertions(+), 51 deletions(-) create mode 100644 include/dt-bindings/sound/fsl-imx-audmux.h diff --git a/include/dt-bindings/sound/fsl-imx-audmux.h b/include/dt-bindings/sound/fsl-imx-audmux.h new file mode 100644 index 0000000..50b09e9 --- /dev/null +++ b/include/dt-bindings/sound/fsl-imx-audmux.h @@ -0,0 +1,56 @@ +#ifndef __DT_FSL_IMX_AUDMUX_H +#define __DT_FSL_IMX_AUDMUX_H + +#define MX27_AUDMUX_HPCR1_SSI0 0 +#define MX27_AUDMUX_HPCR2_SSI1 1 +#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 +#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 +#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 +#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 + +#define MX31_AUDMUX_PORT1_SSI0 0 +#define MX31_AUDMUX_PORT2_SSI1 1 +#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 +#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 +#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 +#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 + +#define MX51_AUDMUX_PORT1_SSI0 0 +#define MX51_AUDMUX_PORT2_SSI1 1 +#define MX51_AUDMUX_PORT3 2 +#define MX51_AUDMUX_PORT4 3 +#define MX51_AUDMUX_PORT5 4 +#define MX51_AUDMUX_PORT6 5 +#define MX51_AUDMUX_PORT7 6 + +/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) +#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) +#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) +#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) +#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) +#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) +#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) +#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) +#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) +#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) + +/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ +#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) +#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) +#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) +#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) +#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) +#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) +#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) +#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) +#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) + +#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) +#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) +#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) + +#endif /* __DT_FSL_IMX_AUDMUX_H */ diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index b8ff44b..38a4209 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -1,57 +1,7 @@ #ifndef __IMX_AUDMUX_H #define __IMX_AUDMUX_H -#define MX27_AUDMUX_HPCR1_SSI0 0 -#define MX27_AUDMUX_HPCR2_SSI1 1 -#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 -#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 -#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 -#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 - -#define MX31_AUDMUX_PORT1_SSI0 0 -#define MX31_AUDMUX_PORT2_SSI1 1 -#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 -#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 -#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 -#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 -#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 - -#define MX51_AUDMUX_PORT1_SSI0 0 -#define MX51_AUDMUX_PORT2_SSI1 1 -#define MX51_AUDMUX_PORT3 2 -#define MX51_AUDMUX_PORT4 3 -#define MX51_AUDMUX_PORT5 4 -#define MX51_AUDMUX_PORT6 5 -#define MX51_AUDMUX_PORT7 6 - -/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) -#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) -#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) -#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) -#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) -#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) -#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) -#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) -#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) -#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) - -/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ -#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) -#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) -#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) -#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) -#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) -#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) -#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) -#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) -#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) - -#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) -#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) -#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) -#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) +#include <dt-bindings/sound/fsl-imx-audmux.h> int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); -- 1.8.3.2

2 1

Re: [alsa-devel] RealTek Inverted Mic Patch
by Takashi Iwai 16 Aug '13

16 Aug '13

At Thu, 15 Aug 2013 14:27:07 -0600, Nathanael D. Noblet wrote: > > Hello, > > I have hardware that suffers from an internal phase inverted mic that > isn't properly detected. Below is a patch that fixes the issue for my > particular hardware. > > Pre patch alsa-info: > > http://www.alsa-project.org/db/?f=a922c77ddf473a8f478ec334fa9dca684f213b59 > > Post patch alsa-info: > http://www.alsa-project.org/db/?f=14cd25b323d3bbb02fabbac78e62851569893c3f > > Incase you care. The patch is small. I don't know if you need any kind > of signed off/written by etc for something so small. In any case there > is no intellectual property in this at all. Feel free to do as you wish > with it. However including it in the kernel would make my life easier. Thanks, I fixed it in sound git tree now. Takashi > > Thanks, > -- > Nathanael d. Noblet > t 403.875.4613 > [2 alsa-realtek-inverted-mic.patch <text/x-patch (7bit)>] > --- a/sound/pci/hda/patch_realtek.c.org 2013-08-15 13:17:49.804492932 -0600 > +++ b/sound/pci/hda/patch_realtek.c 2013-08-15 13:18:40.582364931 -0600 > @@ -4208,6 +4208,7 @@ > SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), > SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68), > SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), > + SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), > > #if 0 > /* Below is a quirk table taken from the old code.

1 0

[alsa-devel] ASoC updates for v3.11
by Mark Brown 15 Aug '13

15 Aug '13

The following changes since commit d4e4ab86bcba5a72779c43dc1459f71fea3d89c8: Linux 3.11-rc5 (2013-08-11 18:04:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc5 for you to fetch changes up to 44ffb69ec617f64cd83182e8b64d93b84758d636: Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus (2013-08-15 11:37:54 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.11 A few driver specific fixes here plus one core fix for a memory corruption issue in DAPM initialisation which could lead to crashes. ---------------------------------------------------------------- Brian Austin (2): ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume ASoC: cs42l52: Add new TLV for Beep Volume Lars-Peter Clausen (1): ASoC: dapm: Fix empty list check in dapm_new_mux() Lothar Waßmann (2): ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control Mark Brown (4): Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus Stephen Warren (1): ASoC: tegra: fix Tegra30 I2S capture parameter setup sound/soc/codecs/cs42l52.c | 5 ++++- sound/soc/codecs/sgtl5000.c | 18 ++++++++++++++---- sound/soc/soc-dapm.c | 7 ++++--- sound/soc/tegra/tegra30_i2s.c | 2 +- 4 files changed, 23 insertions(+), 9 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: core: allow SND_DMAENGINE_PCM use from modules
by Daniel Mack 15 Aug '13

15 Aug '13

When users of SND_DMAENGINE_PCM are built as module, the config symbol SND_DMAENGINE_PCM must be tristate, otherwise the linker will fail. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sound/core/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 94ce1c4..313f22e 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -7,7 +7,7 @@ config SND_PCM select SND_TIMER config SND_DMAENGINE_PCM - bool + tristate config SND_HWDEP tristate -- 1.8.3.1

2 1

Re: [alsa-devel] [PATCH 0/2] add sysfs for acpi interfaces of thinkpad hardware mute
by David Henningsson 15 Aug '13

15 Aug '13

(Added alsa-devel to CC) On 08/14/2013 09:16 AM, Alex Hung wrote: > Added in David and Tim who also has some ideas to share. > > On Wed, Aug 14, 2013 at 3:00 PM, Matthew Garrett > <matthew.garrett(a)nebula.com> wrote: >> On Wed, 2013-08-14 at 14:41 +0800, Alex Hung wrote: >>> Hi Matthew, >>> >>> The problem is that thinkpad-acpi does not aware of mute is changed by >>> either hotkey presses or in desktop. >> >> Ok, how about this. Register the LED with the kernel LED subsystem. Add >> a new LED trigger for volume-mute. Have the HDA driver call that trigger >> whenever the mute state changes. >> >> -- >> Matthew Garrett <matthew.garrett(a)nebula.com> > > > So, to take a wider grip on this issue. First, the hardware mute. Because this is directly connected to the internal speaker (and headphones?), it needs to turn into a ALSA kcontrol on the sound card that controls speaker and/or headphones. And it needs to named accordingly, e g "Speaker Playback Switch" if it controls the speaker only. Exactly how this is going to work out given the arbitrary module load order of snd-hda-intel and thinkpad-acpi, I'm not sure, but the way of creating another sound card (which I've seen on some thinkpads) just isn't working out for userspace. Second, about the mute LEDs and mic-mute LEDs, we currently have several interfaces to choose from: a) HP LEDs currently uses a ALSA kcontrol to control them. For Mute LED there is also a limited automatic control of this LED from kernel space (by default), so it follows the status of "Master Playback Switch" IIRC. b) thinkpad-acpi currently uses custom sysfs nodes to control the LEDs. c) and we also have the /sys/class/leds interface. There might be even more solutions. It's important we get a consistent interfaces towards userspace for the LEDs. And we should also make sure we don't have a permissions problem: if we want the mute/mic-mute LED to follow the status of the currently selected sound card in PulseAudio, which IMO should be our goal, writes cannot be root-only. To have the HDA driver call into the LED subsystem is better than nothing, but in the end we would want the LEDs to follow whatever sound input/output the user currently uses, whether that is bluetooth, USB, HDMI, or internal audio. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 11

[alsa-devel] [PATCH 0/5] ASoC: pxa dmaengine preparation patches
by Daniel Mack 15 Aug '13

15 Aug '13

Mark, This series is an excerpt of my pxa-dma rework that can be merged independently in the next merge window. The rest of the series will take some time to get sorted out, and given that -rc5 is released already, I don't think the entire series will make it in time. Thanks, Daniel Daniel Mack (5): ALSA: move dmaengine implementation from ASoC to ALSA core ASoC: pxa: pxa-ssp: add DT bindings ASoC: pxa: use snd_dmaengine_dai_dma_data ASoC: pxa: pxa-ssp: set dma filter data from startup hook ASoC: pxa: add DT bindings for pxa2xx-pcm .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 ++ .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + include/sound/pxa2xx-lib.h | 7 - sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 ++- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + sound/core/pcm_dmaengine.c | 367 +++++++++++++++++++++ sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/omap/Kconfig | 2 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 8 +- sound/soc/pxa/mmp-sspa.c | 12 +- sound/soc/pxa/pxa-ssp.c | 76 +++-- sound/soc/pxa/pxa2xx-ac97.c | 67 ++-- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/soc-dmaengine-pcm.c | 367 --------------------- sound/soc/spear/Kconfig | 2 +- 22 files changed, 607 insertions(+), 499 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 sound/core/pcm_dmaengine.c delete mode 100644 sound/soc/soc-dmaengine-pcm.c -- 1.8.3.1

3 14

[alsa-devel] [PATCH 1/2] ASoC: pcm3008: Add DAPM support
by Mark Brown 15 Aug '13

15 Aug '13

From: Mark Brown <broonie(a)linaro.org> Make it possible to connect external devices to the CODEC and ensure continued operation with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm3008.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8b9b378..19f5028 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,6 +28,22 @@ #include "pcm3008.h" +static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINL"), +SND_SOC_DAPM_INPUT("VINR"), + +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { + { "PCM3008 Capture", NULL, "VINL" }, + { "PCM3008 Capture", NULL, "VINR" }, + + { "VOUTL", NULL, "PCM3008 Playback" }, + { "VOUTR", NULL, "PCM3008 Playback" }, +}; + #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -77,6 +93,10 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { .suspend = pcm3008_soc_suspend, .resume = pcm3008_soc_resume, + .dapm_widgets = pcm3008_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), + .dapm_routes = pcm3008_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes), }; static int pcm3008_codec_probe(struct platform_device *pdev) -- 1.8.4.rc2

1 1

[alsa-devel] [PATCH] ASoC: tegra: support a Mic Jack in the Tegra+RT5640 machine driver
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> Add a Mic Jack widget to the Tegra+RT5640 machine driver, and document this in the DT binding. This enables the DT to include the Mic Jack in the audio routing table, and hence enables capture of audio, in addition to the previously-working playback. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt | 1 + sound/soc/tegra/tegra_rt5640.c | 1 + 2 files changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt index cba4f88..dc62249 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt @@ -16,6 +16,7 @@ Required properties: * Headphones * Speakers + * Mic Jack - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 08794f9..4511c5a 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -99,6 +99,7 @@ static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), }; static const struct snd_kcontrol_new tegra_rt5640_controls[] = { -- 1.8.1.5

2 1

[alsa-devel] [PATCH] ASoC: tegra: fix Tegra30 I2S capture parameter setup
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> The Tegra30 I2S driver was writing the AHUB interface parameters to the playback path register rather than the capture path register. This caused the capture parameters not to be configured at all, so if capturing using non-HW-default parameters (e.g. 16-bit stereo rather than 8-bit mono) the audio would be corrupted. With this fixed, audio capture from an analog microphone works correctly on the Cardhu board. Cc: stable(a)vger.kernel.org Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d04146c..47565fd04 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); -- 1.8.1.5

2 1

[alsa-devel] [PATCH v4 resent 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 15 Aug '13

15 Aug '13

Changelog: v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 100 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1350 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 223 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1854 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

3 10

[alsa-devel] [PATCH 1/16 v2] sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument
by Julia Lawall 14 Aug '13

14 Aug '13

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the arity of the called function. At the suggestion of Robert Jarzmik, this additionally removes the definition of ARRAY_AND_SIZE, which is not used elsewhere in the file. The semantic match that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,f; @@ f(..., - ARRAY_AND_SIZE(e) + e,ARRAY_SIZE(e) ,...) // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not compiled. v2 removes the definition of ARRAY_AND_SIZE. diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e..bbea778 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -56,8 +56,6 @@ #include "pxa2xx-ac97.h" #include "../codecs/wm9713.h" -#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) - #define AC97_GPIO_PULL 0x58 /* Use GPIO8 for rear speaker amplifier */ @@ -133,10 +131,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG);

2 4

[alsa-devel] [PATCH 2/2] ASoC: wm8960: Fix shared LRCLK support
by Ma Haijun 14 Aug '13

14 Aug '13

Shared LRCLK initialization does not survive wm8960_reset, so place it after the reset. Signed-off-by: Ma Haijun <mahaijuns(a)gmail.com> --- sound/soc/codecs/wm8960.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 368d39f..b606dd3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -990,6 +990,9 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pdata && pdata->shared_lrclk) + snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + /* Latch the update bits */ snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); @@ -1041,7 +1044,6 @@ static const struct regmap_config wm8960_regmap = { static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; @@ -1054,16 +1056,6 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } - i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8960: Fix ADC volume bits
by Ma Haijun 14 Aug '13

14 Aug '13

Signed-off-by: Ma Haijun <mahaijuns(a)gmail.com> --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd..368d39f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -263,8 +263,8 @@ SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), -SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, - 0, 127, 0), +SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC, + 0, 255, 0, adc_tlv), SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", WM8960_BYPASS1, 4, 7, 1, bypass_tlv), -- 1.8.1.2

2 1

[alsa-devel] [PATCH 0/29] simplify use of devm_ioremap_resource
by Julia Lawall 14 Aug '13

14 Aug '13

devm_ioremap_resource often uses the result of a call to platform_get_resource as its last argument. devm_ioremap_resource does appropriate error handling on this argument, so error handling can be removed from the call site. To make the connection between the call to platform_get_resource and the call to devm_ioremap_resource more clear, the former is also moved down to be adjacent to the latter. In many cases, this patch changes the specific error value that is returned on failure of platform_get_resource. The semantic patch that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression pdev,res,n,e,e1; expression ret != 0; identifier l; @@ ( res = platform_get_resource(pdev, IORESOURCE_MEM, n); - if (res == NULL) { ... $goto l;\|return ret;$ } e = devm_ioremap_resource(e1, res); | - res = platform_get_resource(pdev, IORESOURCE_MEM, n); ... when != res - if (res == NULL) { ... $goto l;\|return ret;$ } ... when != res + res = platform_get_resource(pdev, IORESOURCE_MEM, n); e = devm_ioremap_resource(e1, res); ) // </smpl>

2 3

[alsa-devel] HDMI audio is not working on my new desktop
by Alexander E. Patrakov 14 Aug '13

14 Aug '13

Hello. I built a new desktop PC based on the Gigabyte H87N-WIFI motherboard. It has one DVI and two HDMI outputs. However, I cannot get any sound over HDMI. The same OS image works perfectly on my laptop that also has an HDMI output. This is not a pulseaudio problem, as I don't get any sound even if I test with this command: pasuspender -- speaker-test -c2 -D hdmi:0,2 # or, for that matter, 0,0 or 0,1 Alsa-info.sh has uploaded its output to http://www.alsa-project.org/db/?f=1ad7192a15ff4a70b5ddb2579476dbb98adbd929 Please note this error in dmesg, it may be relevant: [ 1.803595] hda_codec: invalid CONNECT_LIST verb 5[1]:0 [ 1.803719] hda_codec: invalid CONNECT_LIST verb 7[1]:0 -- Alexander E. Patrakov

1 0

Re: [alsa-devel] Query on Audio DMA using DMAEngine
by Lars-Peter Clausen 14 Aug '13

14 Aug '13

Added alsa-devel to Cc. On 06/28/2013 05:27 AM, Fernandes, Joel wrote: > Hi Lars, > > Hope you are doing well. > > I am implementing Cyclic DMA support in the EDMA driver that is used by Davinci and now newer TI SoCs. > I am thinking once I am done I can plug it into the snd_dmaengine framework. > > Currently however, the davinci-pcm code directly programs the EDMA. That is what I am working to replace with a single driver and adapt to the snd dmaengine framework. However, once the current code in davinci-pcm uses internal RAM as an intermediate step in the whole DMA process (First data is TX to IRAM from DRAM and then from DRAM to the audio device). > > Do you have any ideas on how we can adapt to the framework, such that we can till use the IRAM? Are there any existing implementations out there that do something similar? Hm, I guess using the snd_dmaengine_pcm helper functions here shouldn't be too hard. Using the generic snd_dmaengine_pcm driver will require some extensions to it though. The mmp platform (pxa/mmp-pcm.c) is also using some kind of on-chip memory, so having support for this in the generic driver certainly makes sense. For the chaining you'd probably have to extend the dmaengine framework, since this kind of interleaved mem-to-mem and mem-to-dev cyclic transfer is currently not possible. I'm wondering though why do you need to copy the data to RAM first, is it not possible to map the IRAM to userspace? - Lars

6 28

[alsa-devel] [PATCH v5 0/1] ASoc: kirkwood: extend the kirkwood audio subsystem for DT usage
by Jean-Francois Moine 14 Aug '13

14 Aug '13

This patch series adds DT support to the kirkwood/mvebu audio subsystem. History v5 - change the DT name to "marvell,mvebu-audio" v4 - change the DT name to "mrvl,mvebu-audio" - remove previous patches 1..3 which have been applied by Russell King v3 - change the DT name from "kirkwood-i2s" to "mvebu-pcm-audio" - apply to linux-next - let NULL as the internal i2s clock for compatibility - change i2s external clock name to "extclk" instead of "external" - cancel previous patch 1/4. Russell's patch should be applied instead. v2 - fix compilation errors -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] [PATCH] ASoC: pcm1681: Add DAPM support
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Provide DAPM for the device, ensuring operation with DAPM required by the core and making it easier to hook up external hardware to it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1681.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 51b1866..651ce09 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -206,6 +206,28 @@ static const struct snd_soc_dai_ops pcm1681_dai_ops = { .digital_mute = pcm1681_digital_mute, }; +static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUT1"), +SND_SOC_DAPM_OUTPUT("VOUT2"), +SND_SOC_DAPM_OUTPUT("VOUT3"), +SND_SOC_DAPM_OUTPUT("VOUT4"), +SND_SOC_DAPM_OUTPUT("VOUT5"), +SND_SOC_DAPM_OUTPUT("VOUT6"), +SND_SOC_DAPM_OUTPUT("VOUT7"), +SND_SOC_DAPM_OUTPUT("VOUT8"), +}; + +static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = { + { "VOUT1", NULL, "Playback" }, + { "VOUT2", NULL, "Playback" }, + { "VOUT3", NULL, "Playback" }, + { "VOUT4", NULL, "Playback" }, + { "VOUT5", NULL, "Playback" }, + { "VOUT6", NULL, "Playback" }, + { "VOUT7", NULL, "Playback" }, + { "VOUT8", NULL, "Playback" }, +}; + static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); static const struct snd_kcontrol_new pcm1681_controls[] = { @@ -258,6 +280,10 @@ static const struct regmap_config pcm1681_regmap = { static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { .controls = pcm1681_controls, .num_controls = ARRAY_SIZE(pcm1681_controls), + .dapm_widgets = pcm1681_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1681_dapm_widgets), + .dapm_routes = pcm1681_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1681_dapm_routes), }; static const struct i2c_device_id pcm1681_i2c_id[] = { -- 1.8.4.rc2

1 0

[alsa-devel] [PATCH 2/3 v2] ASoC: wm8997: Initial CODEC driver
by Charles Keepax 13 Aug '13

13 Aug '13

The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus interfacing, for smartphones, tablets and other portable audio devices based on the Arizona platform. This patch adds the wm8997 CODEC driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 13 +- sound/soc/codecs/wm8997.c | 1187 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + 5 files changed, 1228 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..14d2888 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -120,6 +120,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8996 if I2C + select SND_SOC_WM8997 if MFD_WM8997 select SND_SOC_WM9081 if I2C select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS @@ -143,8 +144,10 @@ config SND_SOC_ARIZONA tristate default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y + default y if SND_SOC_WM8997=y default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m + default m if SND_SOC_WM8997=m config SND_SOC_WM_HUBS tristate @@ -492,6 +495,9 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate +config SND_SOC_WM8997 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..2d327ad 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o +snd-soc-wm8997-objs := wm8997.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o @@ -235,6 +236,7 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o +obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index de62581..c460fca 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -199,9 +199,16 @@ int arizona_init_spk(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); - if (ret != 0) - return ret; + switch (arizona->type) { + case WM8997: + break; + default: + ret = snd_soc_dapm_new_controls(&codec->dapm, + &arizona_spkr, 1); + if (ret != 0) + return ret; + break; + } ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, "Thermal warning", arizona_thermal_warn, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c new file mode 100644 index 0000000..3ba0a98 --- /dev/null +++ b/sound/soc/codecs/wm8997.c @@ -0,0 +1,1187 @@ +/* + * wm8997.c -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <linux/mfd/arizona/core.h> +#include <linux/mfd/arizona/registers.h> + +#include "arizona.h" +#include "wm8997.h" + +struct wm8997_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +static const struct reg_default wm8997_sysclk_reva_patch[] = { + { 0x301D, 0x7B15 }, + { 0x301B, 0x0050 }, + { 0x305D, 0x7B17 }, + { 0x305B, 0x0050 }, + { 0x3001, 0x08FE }, + { 0x3003, 0x00F4 }, + { 0x3041, 0x08FF }, + { 0x3043, 0x0005 }, + { 0x3020, 0x0225 }, + { 0x3021, 0x0A00 }, + { 0x3022, 0xE24D }, + { 0x3023, 0x0800 }, + { 0x3024, 0xE24D }, + { 0x3025, 0xF000 }, + { 0x3060, 0x0226 }, + { 0x3061, 0x0A00 }, + { 0x3062, 0xE252 }, + { 0x3063, 0x0800 }, + { 0x3064, 0xE252 }, + { 0x3065, 0xF000 }, + { 0x3116, 0x022B }, + { 0x3117, 0xFA00 }, + { 0x3110, 0x246C }, + { 0x3111, 0x0A03 }, + { 0x3112, 0x246E }, + { 0x3113, 0x0A03 }, + { 0x3114, 0x2470 }, + { 0x3115, 0x0A03 }, + { 0x3126, 0x246C }, + { 0x3127, 0x0A02 }, + { 0x3128, 0x246E }, + { 0x3129, 0x0A02 }, + { 0x312A, 0x2470 }, + { 0x312B, 0xFA02 }, + { 0x3125, 0x0800 }, +}; + +static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm8997_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch); + break; + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + default: + break; + } + + return 0; +} + +static const char *wm8997_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm8997_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm8997_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), +}; + +#define WM8997_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static const struct snd_kcontrol_new wm8997_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, + ARIZONA_EQ1_ENA_MASK), +SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, + ARIZONA_EQ2_ENA_MASK), +SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, + ARIZONA_EQ3_ENA_MASK), +SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, + ARIZONA_EQ4_ENA_MASK), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L), +WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + +static const char *wm8997_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm8997_aec_loopback_values[] = { + 0, 1, 4, 6, 8, 9, +}; + +static const struct soc_enum wm8997_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(wm8997_aec_loopback_texts), + wm8997_aec_loopback_texts, + wm8997_aec_loopback_values); + +static const struct snd_kcontrol_new wm8997_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback); + +static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8997_aec_loopback_mux), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" } + +static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, + { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, + + { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, + { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, + + { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, + { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, + + { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, + { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MUX_ROUTES("ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2"), + + ARIZONA_MUX_ROUTES("ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2"), + + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "AEC Loopback", "EPOUT", "OUT3L" }, + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "AEC Loopback", "SPKOUT", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "MICSUPP", NULL, "SYSCLK" }, +}; + +static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM8997_FLL1: + return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout); + case WM8997_FLL2: + return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout); + case WM8997_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref, + Fout); + case WM8997_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm8997_dai[] = { + { + .name = "wm8997-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-slim1", + .id = 3, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim2", + .id = 4, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim3", + .id = 5, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, +}; + +static int wm8997_codec_probe(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = priv->core.arizona->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + if (ret != 0) + return ret; + + arizona_init_spk(codec); + + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); + + priv->core.arizona->dapm = &codec->dapm; + + return 0; +} + +static int wm8997_codec_remove(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define WM8997_DIG_VU 0x0200 + +static unsigned int wm8997_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8997 = { + .probe = wm8997_codec_probe, + .remove = wm8997_codec_remove, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm8997_set_fll, + + .controls = wm8997_snd_controls, + .num_controls = ARRAY_SIZE(wm8997_snd_controls), + .dapm_widgets = wm8997_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets), + .dapm_routes = wm8997_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes), +}; + +static int wm8997_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm8997_priv *wm8997; + int i; + + wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv), + GFP_KERNEL); + if (wm8997 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm8997); + + wm8997->core.arizona = arizona; + wm8997->core.num_inputs = 4; + + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) + wm8997->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm8997->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm8997->fll[1]); + + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + + for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++) + arizona_init_dai(&wm8997->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], + WM8997_DIG_VU, WM8997_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, + wm8997_dai, ARRAY_SIZE(wm8997_dai)); +} + +static int wm8997_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm8997_codec_driver = { + .driver = { + .name = "wm8997-codec", + .owner = THIS_MODULE, + }, + .probe = wm8997_probe, + .remove = wm8997_remove, +}; + +module_platform_driver(wm8997_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8997 driver"); +MODULE_AUTHOR("Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8997-codec"); diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h new file mode 100644 index 0000000..5e91c6a --- /dev/null +++ b/sound/soc/codecs/wm8997.h @@ -0,0 +1,23 @@ +/* + * wm8997.h -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8997_H +#define _WM8997_H + +#include "arizona.h" + +#define WM8997_FLL1 1 +#define WM8997_FLL2 2 +#define WM8997_FLL1_REFCLK 3 +#define WM8997_FLL2_REFCLK 4 + +#endif -- 1.7.2.5

2 3

[alsa-devel] [PATCH 1/3] ASoC: max9877: Convert to use regmap API
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max9877.c | 187 ++++++++++++++++++++++++++++----------------- 1 file changed, 118 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 6b6c74c..7e2fe50 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -14,27 +14,21 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <sound/soc.h> #include <sound/tlv.h> #include "max9877.h" -static struct i2c_client *i2c; +static struct regmap *regmap; -static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; - -static void max9877_write_regs(void) -{ - unsigned int i; - u8 data[6]; - - data[0] = MAX9877_INPUT_MODE; - for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) - data[i + 1] = max9877_regs[i]; - - if (i2c_master_send(i2c, data, 6) != 6) - dev_err(&i2c->dev, "i2c write failed\n"); -} +static struct reg_default max9877_regs[] = { + { 0, 0x40 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x49 }, +}; static int max9877_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -45,8 +39,14 @@ static int max9877_get_reg(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int mask = mc->max; unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = @@ -65,18 +65,21 @@ static int max9877_set_reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int invert = mc->invert; unsigned int val = (ucontrol->value.integer.value[0] & mask); + bool change; + int ret; if (invert) val = mask - val; - if (((max9877_regs[reg] >> shift) & mask) == val) - return 0; - - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_write_regs(); + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change); + if (ret != 0) + return ret; - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_2reg(struct snd_kcontrol *kcontrol, @@ -88,9 +91,18 @@ static int max9877_get_2reg(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int mask = mc->max; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[0] = (val >> shift) & mask; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + ret = regmap_read(regmap, reg2, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[1] = (val >> shift) & mask; return 0; } @@ -106,77 +118,99 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 0; - - if (((max9877_regs[reg] >> shift) & mask) != val) - change = 1; - - if (((max9877_regs[reg2] >> shift) & mask) != val2) - change = 1; - - if (change) { - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= val2 << shift; - max9877_write_regs(); - } - - return change; + bool change1, change2; + int ret; + + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change1); + if (ret != 0) + return ret; + + ret = regmap_update_bits_check(regmap, reg2, mask << shift, + val2 << shift, &change2); + if (ret != 0) + return ret; + + if (change1 || change2) + return 1; + else + return 0; } static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OUTMODE_MASK; + if (val) + val--; - if (value) - value -= 1; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; + unsigned int val; + bool change; + int ret; - value += 1; + val = ucontrol->value.integer.value[0] + 1; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) - return 0; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OUTMODE_MASK, val, &change); + if (ret != 0) + return ret; - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OSC_MASK; + val >>= MAX9877_OSC_OFFSET; - value = value >> MAX9877_OSC_OFFSET; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; - - value = value << MAX9877_OSC_OFFSET; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + unsigned int val; + bool change; + int ret; + + val = ucontrol->value.integer.value[0] << MAX9877_OSC_OFFSET; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OSC_MASK, val, &change); + if (ret != 0) + return ret; + + if (change) + return 1; + else return 0; - - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; } static const unsigned int max9877_pgain_tlv[] = { @@ -258,19 +292,34 @@ int max9877_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct regmap_config max9877_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max9877_regs, + .num_reg_defaults = ARRAY_SIZE(max9877_regs), + .cache_type = REGCACHE_RBTREE, +}; + static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - i2c = client; + int i; + + regmap = devm_regmap_init_i2c(client, &max9877_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - max9877_write_regs(); + /* Ensure the device is in reset state */ + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); return 0; } static int max9877_i2c_remove(struct i2c_client *client) { - i2c = NULL; + regmap = NULL; return 0; } -- 1.8.4.rc2

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[alsa-devel] [PATCH v2] extcon: arizona: Clear trig_sts bits on all paths
by Charles Keepax 13 Aug '13

13 Aug '13

We want the trig_sts bits to be cleared in all cases where we consider the jack detection interrupt to have been handled. Specifically, if a duplicate detection event was suppressed these bits were not cleared causing the CODEC to not enter a low power state. This patch clears the bits on the duplicate detection code path. Reported-by: Ryo Tsutsui <ryo.tsutsui(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/extcon/extcon-arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index 7a1b4a7..4df68de 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -1034,6 +1034,7 @@ static irqreturn_t arizona_jackdet(int irq, void *data) else info->micd_timeout = DEFAULT_MICD_TIMEOUT; +out: /* Clear trig_sts to make sure DCVDD is not forced up */ regmap_write(arizona->regmap, ARIZONA_AOD_WKUP_AND_TRIG, ARIZONA_MICD_CLAMP_FALL_TRIG_STS | @@ -1041,7 +1042,6 @@ static irqreturn_t arizona_jackdet(int irq, void *data) ARIZONA_JD1_FALL_TRIG_STS | ARIZONA_JD1_RISE_TRIG_STS); -out: mutex_unlock(&info->lock); pm_runtime_mark_last_busy(info->dev); -- 1.7.2.5

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[alsa-devel] [PATCH] Add PM support to the rme96 driver
by Knut Petersen 13 Aug '13

13 Aug '13

Hi everybody! The attached patch adds PM support to the rme96 driver, it should be applied after v2 of the patch adding pcm stream synchronization support posted on 2013-08-08. It does work perfectly here on an RME Digi96/8 PAD: After connecting analog output and analog input executing #! /bin/sh # arecord -D hw:0,0 -d 62 testout.wav & aplay -D hw:0,0 -d 60 testin.wav produces an almost perfect recording even when the test machine is suspended / hibernated and resumed several times. cu, Knut

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[alsa-devel] [PATCH] ASoC: dapm: Ensure kcontrol list is initialised
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Ensure that the recently added path kcontrol list is initialised otherwise we may crash trying to delete routes that don't have kcontrols. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b78eaa1..1e0e4ce 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2357,6 +2357,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, path->sink = wsink; path->connected = connected; INIT_LIST_HEAD(&path->list); + INIT_LIST_HEAD(&path->list_kcontrol); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); -- 1.8.4.rc1

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[alsa-devel] [PATCH] ASoC: dapm: Don't create routes when creating kcontrols
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> Attempting to create the route as part of adding a mux control causes us to attempt to add the same route twice since we loop over all sources for the mux after creating the control. Instead do the addition in the callers. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-dapm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 662a904..b78eaa1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -665,7 +665,7 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, * create it. Either way, add the widget into the control's widget list */ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, - int kci, struct snd_soc_dapm_path *path) + int kci) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; @@ -766,7 +766,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, return ret; w->kcontrols[kci] = kcontrol; - dapm_kcontrol_add_path(kcontrol, path); return 0; } @@ -790,9 +789,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) continue; } - ret = dapm_create_or_share_mixmux_kcontrol(w, i, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, i); if (ret < 0) return ret; + + dapm_kcontrol_add_path(w->kcontrols[i], path); } } @@ -821,7 +822,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) path = list_first_entry(&w->sources, struct snd_soc_dapm_path, list_sink); - ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0); if (ret < 0) return ret; -- 1.8.4.rc1

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[alsa-devel] [PATCH 1/2] ALSA: hda - Allow auto_mute_via_amp on bind mute controls
by Takashi Iwai 13 Aug '13

13 Aug '13

The auto-mute using the amp currently works only for a single amp on a pin. Make it working also with HDA_CTL_BIND_MUTE type, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f6c0344..6ed2209 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -816,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx) static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); enum { HDA_CTL_WIDGET_VOL, @@ -833,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = { .put = hda_gen_mixer_mute_put, /* replaced */ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), }, - HDA_BIND_MUTE(NULL, 0, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_bind_switch_get, + .put = hda_gen_bind_mute_put, /* replaced */ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), + }, }; /* add dynamic controls from template */ @@ -940,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx, } /* playback mute control with the software mute bit check */ -static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gen_spec *spec = codec->spec; @@ -952,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] &= enabled; ucontrol->value.integer.value[1] &= enabled; } +} +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); + return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol); +} + /* any ctl assigned to the path with the given index? */ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) { -- 1.8.3.1

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[alsa-devel] [PATCH 1/3] ASoC: wm5102: Add inputs for noise and mic mixers
by Mark Brown 13 Aug '13

13 Aug '13

From: Mark Brown <broonie(a)linaro.org> The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f38c52d..8bbddc1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1423,9 +1423,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -1552,6 +1549,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), -- 1.8.4.rc1

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[alsa-devel] [PATCH v2 0/2] ALSA: Implement core jack support for kcontrol
by Felipe F. Tonello 12 Aug '13

12 Aug '13

From: "Felipe F. Tonello" <eu(a)felipetonello.com> On this second version I made sure that the patch series can be bisected. Also, since HDA codec has it's own jack infrastructure, I just disabled the support of the core ALSA jack in it. Why? Because it's causing duplication of controls, although the input events works fine. A HDA codec developer might give us more information about it. Patch 1) Is the actual implementation and fixes for users that uses the API. Patch 2) Name fixes. Just to make sure no jack has " Jack" duplicated at the end of its name. Felipe F. Tonello (2): ALSA: Added jack detection KControl support ALSA: SoC: Fix jack names according new API include/sound/control.h | 2 +- include/sound/jack.h | 8 ++-- include/sound/soc.h | 2 +- sound/core/Kconfig | 1 + sound/core/ctljack.c | 15 ++++++- sound/core/jack.c | 88 +++++++++++++++++++++++++++++++++++--- sound/pci/hda/Kconfig | 8 ---- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 4 +- sound/soc/mid-x86/mfld_machine.c | 6 +-- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 4 +- sound/soc/omap/omap-twl4030.c | 4 +- sound/soc/omap/rx51.c | 6 +-- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/palm27x.c | 4 +- sound/soc/pxa/ttc-dkb.c | 10 ++--- sound/soc/pxa/z2.c | 4 +- sound/soc/samsung/goni_wm8994.c | 7 +-- sound/soc/samsung/h1940_uda1380.c | 4 +- sound/soc/samsung/littlemill.c | 10 ++--- sound/soc/samsung/lowland.c | 6 +-- sound/soc/samsung/rx1950_uda1380.c | 4 +- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/speyside.c | 6 +-- sound/soc/samsung/tobermory.c | 4 +- sound/soc/soc-jack.c | 5 ++- sound/soc/tegra/tegra_alc5632.c | 4 +- sound/soc/tegra/tegra_rt5640.c | 4 +- sound/soc/tegra/tegra_wm8903.c | 8 ++-- 30 files changed, 163 insertions(+), 77 deletions(-) -- 1.8.3.1

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[alsa-devel] [RFC PATCH] ASoC: omap-mcbsp: Add support for highter rate up to 384000
by Michael Trimarchi 12 Aug '13

12 Aug '13

Add support for highter rate up to 384000Khz. There are codecs in the market that can play up to this frequency. Test was done just with: #define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000) playing wav file 32bit/192000Khz stereo. Some glitch has observed but this should depend on the selected min buffer bytes that was left untouched in aplay and in the omap-pcm driver. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..1e95f6a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,7 +39,9 @@ #include "mcbsp.h" #include "omap-mcbsp.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000 | \ + SNDRV_PCM_RATE_CONTINUOUS | \ + SNDRV_PCM_RATE_KNOT) #define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ xhandler_get, xhandler_put) \ @@ -574,11 +576,15 @@ static struct snd_soc_dai_driver omap_mcbsp_dai = { .channels_max = 16, .rates = OMAP_MCBSP_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 384000, }, .capture = { .channels_min = 1, .channels_max = 16, .rates = OMAP_MCBSP_RATES, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &mcbsp_dai_ops, -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: dt: Move ALC5632 pin list from Tegra board binding to CODEC binding
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../devicetree/bindings/sound/alc5632.txt | 20 ++++++++++++++++++ .../bindings/sound/nvidia,tegra-audio-alc5632.txt | 24 ++-------------------- 2 files changed, 22 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt index 8608f74..b913b74 100644 --- a/Documentation/devicetree/bindings/sound/alc5632.txt +++ b/Documentation/devicetree/bindings/sound/alc5632.txt @@ -13,6 +13,26 @@ Required properties: - #gpio-cells : Should be two. The first cell is the pin number and the second cell is used to specify optional parameters (currently unused). +Pins on the device (for linking into audio routes): + + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + * DMICDAT + + Example: alc5632: alc5632@1e { diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt index 05ffecb..8b8903e 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt @@ -11,28 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the ALC5632's pins: - - ALC5632 pins: - - * SPK_OUTP - * SPK_OUTN - * HP_OUT_L - * HP_OUT_R - * AUX_OUT_P - * AUX_OUT_N - * LINE_IN_L - * LINE_IN_R - * PHONE_P - * PHONE_N - * MIC1_P - * MIC1_N - * MIC2_P - * MIC2_N - * MICBIAS1 - * DMICDAT - - Board connectors: + sinks are the ALC5632's pins as documented in the binding for the device + and: * Headset Stereophone * Int Spk -- 1.8.4.rc1

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Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Daniel Mack 12 Aug '13

12 Aug '13

On 12.08.2013 13:27, Jesus R wrote: > On Mon, Aug 12, 2013 at 4:52 AM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > Right now MPD supports DSD/DoP with one selectable on/off option. On > equals DoP output and off equals conversion to PCM. I'm not aware of any > case where it does not work. I would seem to me that there is one > predominant format? Let me clarify that again: I read the above as the two options are a) DSD material is augmented by DOP markers and send as such over a regular PCM interface, or b) convert DSD to PCM in software and then send PCM samples over a regular PCM interface. What applications need to use for my new approach is another is: c) take DSD samples, bit reverse them if necessary, and then send them over the generic DSD data pipe to the Linux kernel. Device-specific details are taken care of by the kernel layer. I'm not aware of any software that uses that new DSD format type directly except for my example program. Daniel

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Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Daniel Mack 12 Aug '13

12 Aug '13

[please never drop people from the Cc: list but always use 'reply to all'!] On 12.08.2013 13:27, Jesus R wrote: > On Mon, Aug 12, 2013 at 4:52 AM, Daniel Mack <zonque(a)gmail.com > <mailto:zonque@gmail.com>> wrote: > Given than none of devices have a chance to report their capabilities to > the host, finding out which modes to use is a trial-and-error game, > unless you have access to the firmware sources. > > > Hence, please just look up the USB IDs and alternate interface settings > of your connected device with lsusb, and then augment the code in > snd_usb_interface_dsd_format_quirks() (file sound/usb/quirks.c) and play > around with different possibilities (bitreverse on/off, DSD_U8 vs > DSD_U16 etc). Follow the example that I provided in that function. > > Right now MPD supports DSD/DoP with one selectable on/off option. On > equals DoP output and off equals conversion to PCM. I'm not aware of any > case where it does not work. I would seem to me that there is one > predominant format? It's more complicated than that. Applications were always free to use a usual PCM stream and stuff it with DOP/DSD samples. This is the whole idea of DOP: allow applications to transport DSD samples via regular audio streams. The downside of this is that applications are in charge of adding the DOP markers manually, and information about bit-reversing and marker format (8 bit vs. 16 bit) is up to the application, which is not the right place for it. The patch series I brought in addressed this in a way that all that information about device-specific details is now known to the kernel, which can hence offer a standard compliant interface to userspace. [There's yet one more level of confusion as some file formats store the samples bit-reversed, but there's nothing we can do about that from kernel space except for defining a stable interface application have to obey to in order to be compatible. That is, samples are expected to always carry the oldest bit in the most significant bit of a byte.] So my question is: are you 100% sure that MPD uses the newly introduces ALSA formats, or are they doing the DOP conversion in manually in their own software layers? > You have to recompile and rmmod/insmod the snd-usb driver every time you > make a change, and then you need some sort of userspace tool to test it. > I published one here: > > This may be more than I can do. Can you add the options (bitreverse > on/off, DSD_U8 vs > DSD_U16 etc) to the alas-dsd-player? This would allow me to quickly test > the devices. There are #if0'ed blocks that you can opt-in for bit reversal, but as the marker format is different for 8 and 16 bits, and up to the kernel space now, the dsd player does not know about them. That was the entire intention of bringing that logic to kernel space in the first place. > Take a look at the DSD database so you can see what we are up against. I > could even add the tests results to the database. > https://docs.google.com/spreadsheet/ccc?key=0AgVhKcl_3lHfdFVyenBBNjNpQ2lieG… Nice list. Only devics with USB interface are interesting at this point, but that's a good start. As I said: you need to add the quirks for the devices to the quirks list. Other than that, there's no chance to test them. Copy'n pasting code inside that function and recompiling the kernel is really not as hard as you might think it is. Just search the internet for "compile kernel git <your-distribution-here>" and follow the links. Thanks, Daniel

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[alsa-devel] [PATCH V4 0/4] Add i2s support on smdk5420
by Padmavathi Venna 12 Aug '13

12 Aug '13

Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control, internal mux for rclk src clk and tdm support. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. This also adds i2s support on exynos5420. Changes since V3: - Addressed review comments by Tomasz Figa related to const qualifier for samsung_i2s_dai_data - Removed passing quirks as driver data for non-dt platforms. - Separated out adding i2s nodes and enabling audio support on 5420 into different patch set as they are dependent on some of already posted but not yet merged i2c, dwmmc, dma and audss clock controller patches. Changes since V2: - Separated out driver side changes and dts changes in two patch sets. - Replaced samsung,s3c6410-i2s-v4 with samsung,s3c6410-i2s-multi for more clarity as suggested by Tomasz Figa. Changes since V1: - Pass quirks as driver data and fetch the quirks based on compatible string from dtsi file as suggested by Tomasz Figa and Mark Brown - Make the I2S driver more flexible with respect to register access as suggested by Tomasz Figa and Mark Brown - Add 5420 support in the driver. - Modify the dtsi files with the corresponding compatible strings and removed the i2s quirks from 5250 dtsi file. - Updated the i2s Documentation with relevent changes and i2s versioning info. - Add i2s nodes on exynos5420.dtsi - Enable sound support on smdk5420 This patch set is made based on Mark Brown for-next branch on sound.git. Padmavathi Venna (4): ASoC: Samsung: I2S: Add quirks as driver data in I2S ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 ARM: dts: exynos5250: move common i2s properties to exynos5 dtsi ARM: dts: Change i2s compatible string on exynos5250 .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++-- arch/arm/boot/dts/exynos5.dtsi | 21 +++ arch/arm/boot/dts/exynos5250.dtsi | 17 +--- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 15 ++ sound/soc/samsung/i2s.c | 143 +++++++++++++++----- 6 files changed, 157 insertions(+), 62 deletions(-) -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: arizona: Add widget<->mux route into mux route macro
by Charles Keepax 12 Aug '13

12 Aug '13

The routes linking the widget and the input mux were being added manually, rather than by the ARIZONA_MUX_ROUTES macro. This patchs adds the routes to the macro. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 3 ++- sound/soc/codecs/wm5102.c | 41 ++++++++++++----------------------------- sound/soc/codecs/wm5110.c | 8 ++++---- 3 files changed, 18 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b60b08c..bf67d65 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -150,7 +150,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \ ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux) -#define ARIZONA_MUX_ROUTES(name) \ +#define ARIZONA_MUX_ROUTES(widget, name) \ + { widget, NULL, name " Input" }, \ ARIZONA_MIXER_INPUT_ROUTES(name " Input") #define ARIZONA_MIXER_ROUTES(widget, name) \ diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 282fd23..9e9a0c1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1499,23 +1499,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L PGA", NULL, "IN3L" }, { "IN3R PGA", NULL, "IN3R" }, - { "ASRC1L", NULL, "ASRC1L Input" }, - { "ASRC1R", NULL, "ASRC1R Input" }, - { "ASRC2L", NULL, "ASRC2L Input" }, - { "ASRC2R", NULL, "ASRC2R Input" }, - - { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, - { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, - - { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, - { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, - - { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, - { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, - - { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, - { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, - ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -1567,22 +1550,22 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), - ARIZONA_MUX_ROUTES("ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), - ARIZONA_MUX_ROUTES("ISRC1DEC1"), - ARIZONA_MUX_ROUTES("ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), - ARIZONA_MUX_ROUTES("ISRC2INT1"), - ARIZONA_MUX_ROUTES("ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), - ARIZONA_MUX_ROUTES("ISRC2DEC1"), - ARIZONA_MUX_ROUTES("ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), ARIZONA_DSP_ROUTES("DSP1"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e7cb4b..0cfc5e6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -979,10 +979,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, -- 1.7.2.5

2 1

[alsa-devel] Another possible race in deferred probing...
by Russell King - ARM Linux 12 Aug '13

12 Aug '13

There is no better way to provoke bugs than to show a device to someone who has never seen it before. Obviously, I didn't get much chance to investigate what happened, so all I have are the kernel messages: kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CPU DAI mvebu-audio.1 not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral mvebu-audio mvebu-audio.1: found external clock kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CODEC spdif-dit not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral kirkwood-spdif-audio kirkwood-spdif-audio.1: ASoC: CODEC spdif-dit not registered kirkwood-spdif-audio kirkwood-spdif-audio.1: failed to register card platform kirkwood-spdif-audio.1: Driver kirkwood-spdif-audio requests probe deferral and then when manually removing the kirkwood-spdif module and re-inserting it: kirkwood-spdif-audio kirkwood-spdif-audio.1: dit-hifi <-> mvebu-audio.1 mapping ok First, remember that Ubuntu loads kernel modules from separate threads, so module loading here is multithreaded. What I suspect happened was the spdif-dit module was inserted and initialized between kirkwood-spdif starting to be re-probed, and returning from its probe function with -EPROBE_DEFER, and I suspect that spdif-dit was the last module. As there was no further module loading or driver activity, there was no attempt to re-probe the kirkwood-spdif module. This seems to be a very rare event (it's the first time I've seen it since the last fix to the deferred probing, and that's many hundreds of reboots.) Nevertheless, the fact that it has happened indicates that deferred probing still has holes that can leave modular drivers unbound after boot. Let's consider this scenario, which from a quick scan of the code looks like it's possible to happen: Thread 0 Thread 1 deferred probe triggered driver A's probe function called (kirkwood_spdif_probe in my case) takes lock L to lookup resource A checks for resource A (the spdif_dit codec) releases lock L resource A is not found ----- context switch ------ module init function called (to register spdif_dit_driver) driver B's probe function called (to register the spdif_dit codec via snd_soc_register_codec) takes lock L for adding resource A registers resource A releases lock L driver B's probe function completes deferred probe triggered, deferred probe list empty module init completes ----- context switch ------ probe returns -EPROBE_DEFER deferred probe mutex taken added to deferred probe list deferred probe mutex released This can happen because there's no locking between looking up the resource and having the driver added to the list of devices waiting to be probed. Given that deferred probing is becoming the "normal" thing for device drivers, these races are only going to get worse. Consider that as we're working on ARM towards having a single zImage bootable on many boards, most drivers will be modules, and if they're inserted into the kernel in a multi-threaded way, the above race is going to continually come back and bite us. I don't see an easy solution to this; we can't serialise device probes, because we have some places where device driver probe functions register devices themselves, which then go on to have their drivers probed (nested probes). Maybe this reflects back into Greg's whole argument against platform devices: maybe we need a struct device which lists _all_ the resources which a device requires, have the bus layer safely check for the presence of all those resources before calling the probe function, and move the deferred probe out of the core into the bus layer(s). Another solution would be to have a way to attach to the device a list of resources which failed, and have the driver core scan that list after the probe returns rechecking to see if all the resources are now present. That to me sounds particularly horrid, invasive and quite disgusting to implement safely.

1 0

[alsa-devel] [PATCH] ALSA: compress: update the MAINTAINER entry
by Vinod Koul 12 Aug '13

12 Aug '13

add missing uapi/ headers and Documentation files Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- MAINTAINERS | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 38a870e..a40e393 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7667,7 +7667,9 @@ M: Vinod Koul <vinod.koul(a)intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Supported +F: Documentation/sound/alsa/compress_offload.txt F: include/sound/compress_driver.h +F: include/uapi/sound/compress_* F: sound/core/compress_offload.c F: sound/soc/soc-compress.c -- 1.7.0.4

2 1

[alsa-devel] [PATCH 2/2] 6fire: make buffers DMA-able (midi)
by Torsten Schenk 12 Aug '13

12 Aug '13

Patch makes midi output buffer DMA-able by allocating it separately. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> --- diff -Nur a/sound/6fire/midi.c b/sound/6fire/midi.c --- a/sound/6fire/midi.c 2013-08-04 22:46:46.000000000 +0200 +++ b/sound/6fire/midi.c 2013-08-11 10:51:05.682482388 +0200 @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff -Nur a/sound/6fire/midi.h b/sound/6fire/midi.h --- a/sound/6fire/midi.h 2013-08-04 22:46:46.000000000 +0200 +++ b/sound/6fire/midi.h 2013-08-11 10:51:05.682482388 +0200 @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length);

2 1

[alsa-devel] [PATCH] 6fire: fix URB transfer buffer for midi output
by Torsten Schenk 12 Aug '13

12 Aug '13

Patch fixes URB transfer buffer allocation for midi output to be DMA-able. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> --- diff -Nur a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c --- a/sound/usb/6fire/midi.c 2013-08-07 16:32:10.579639391 +0200 +++ b/sound/usb/6fire/midi.c 2013-08-07 16:32:31.363378104 +0200 @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff -Nur a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h --- a/sound/usb/6fire/midi.h 2013-08-07 16:32:10.579639391 +0200 +++ b/sound/usb/6fire/midi.h 2013-08-07 16:32:31.363378104 +0200 @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length);

4 7

[alsa-devel] LG LW25 pin fix
by Joonas Saarinen 12 Aug '13

12 Aug '13

Hello, I discovered that the LG LW25 laptop needs a small fix to its pin mapping. The functions of blue and green jacks simply have to be interchanged. The rest of the configuration is correct. Can this fix be made permanent in the driver? The fixed mappings are: 0x1a 0x0181344f 0x1b 0x0321403f Here's the full correct map: [codec] 0x10ec0880 0x00001854 0 [pincfg] 0x14 0xb7031110 0x15 0x411111f0 0x16 0x411111f0 0x17 0x411111f0 0x18 0x01a19c22 0x19 0xb7a31120 0x1a 0x0181344f 0x1b 0x0321403f 0x1c 0x411111f0 0x1d 0x411111f0 0x1e 0x0145311e 0x1f 0x411111f0 Thanks, Joonas

2 5

Re: [alsa-devel] [PATCH v3 5/5] ALSA: snd-usb: add quirks handler for DSD streams
by Jesus R 12 Aug '13

12 Aug '13

Daniel/team, I have access to a few DSD devices and or USB based devices. I can provide any information you need to help with allowing ALSA to identify the device's capabilities. Just let me know how to get the information you need from command line:) I also have a list of known DSD/DoP devices in Google Drive that may be helpful. Let me know if it's okay to post the link. Jesus R sonore.us

2 2

[alsa-devel] [PATCH] ASoC: adau1701: Use gpio_set_value_cansleep()
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> The GPIO manipulation done by this driver is never in atomic context so we can use gpio_set_value_cansleep() and support GPIOs that can't be set from atomic context. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/adau1701.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2c10252..ebff112 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) gpio_is_valid(adau1701->gpio_pll_mode[1])) { switch (clkdiv) { case 64: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 256: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; case 384: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 0: /* fallback */ case 512: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; } } @@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) adau1701->pll_clkdiv = clkdiv; if (gpio_is_valid(adau1701->gpio_nreset)) { - gpio_set_value(adau1701->gpio_nreset, 0); + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ udelay(1); - gpio_set_value(adau1701->gpio_nreset, 1); + gpio_set_value_cansleep(adau1701->gpio_nreset, 1); /* power-up time may be as long as 85ms */ mdelay(85); } -- 1.8.4.rc1

2 1

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 12 Aug '13

12 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Please ignore the previous patch. I have tested speaker and headphone playback on this patch. --- sound/soc/codecs/rt5640.c | 389 +++++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5640.h | 15 ++ 2 files changed, 352 insertions(+), 52 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..cae3f46 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -378,21 +376,119 @@ static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; +static const char * const rt5640_mute_mode[] = {"mute", "unmute"}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_hp_enum, 0, 0, rt5640_mute_mode); +static const SOC_ENUM_SINGLE_DECL(rt5640_spo_enum, 0, 0, rt5640_mute_mode); + static const SOC_ENUM_SINGLE_DECL( rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +static int rt5640_spo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->spo_l_mute; + ucontrol->value.integer.value[1] = !rt5640->spo_r_mute; + + return 0; +} + +static int rt5640_spo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_DIG1) & RT5640_PWR_CLS_D; + + if (ucontrol->value.integer.value[0]) { + rt5640->spo_l_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->spo_l_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, + RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->spo_r_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->spo_r_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + } + + return 0; +} + +static int rt5640_hp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->hp_l_mute; + ucontrol->value.integer.value[1] = !rt5640->hp_r_mute; + + return 0; +} + +static int rt5640_hp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + + if (ucontrol->value.integer.value[0]) { + rt5640->hp_l_mute = false; + if (pow & RT5640_PWR_HP_L) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->hp_l_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->hp_r_mute = false; + if (pow & RT5640_PWR_HP_R) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->hp_r_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, RT5640_R_MUTE); + } + + + return 0; +} + static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("Speaker Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_spo_get, rt5640_spo_put), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("HP Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_hp_get, rt5640_hp_put), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -868,33 +964,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +1012,216 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_spo_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_r_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_spo_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_l_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, + RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | + RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(20000, 25000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, + RT5640_PWR_FV1 | RT5640_PWR_FV2); + + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CO_MASK | + RT5640_HP_SG_MASK, + RT5640_HP_CO_EN | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, pow; + + regmap_read(rt5640->regmap, RT5640_PWR_ANLG1, &pow); + if (rt5640->hp_l_mute || !(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (rt5640->hp_r_mute || !(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + /* headphone unmute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK, RT5640_SMT_TRIG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK, RT5640_RSTN_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_RSTN_DIS | RT5640_HP_L_SMT_EN | + RT5640_HP_R_SMT_EN); + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(40000, 45000); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | RT5640_HP_R_SMT_DIS); + +} + +static void rt5640_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* headphone mute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK, RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK, RT5640_RSTP_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_RSTP_DIS | + RT5640_HP_L_SMT_EN | RT5640_HP_R_SMT_EN); + + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + usleep_range(40000, 45000); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | + RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | + RT5640_HP_R_SMT_DIS); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK | + RT5640_HP_CD_PD_MASK | + RT5640_HP_CO_MASK | + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_SMT_TRIG_DIS | + RT5640_HP_CD_PD_EN | + RT5640_HP_CO_DIS | RT5640_HP_CP_PD | + RT5640_HP_SG_EN | RT5640_HP_CB_PD); +} + +static int rt5640_hp_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: + rt5640_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_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: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1411,23 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("SPO L Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_l_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("SPO R Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_r_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1396,13 +1683,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"SPO L Amp", NULL, "SPOL MIX"}, + {"SPO R Amp", NULL, "SPOR MIX"}, + {"SPOLP", NULL, "SPO L Amp"}, + {"SPOLN", NULL, "SPO L Amp"}, + {"SPORP", NULL, "SPO R Amp"}, + {"SPORN", NULL, "SPO R Amp"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,7 +1701,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, + {"HPOL", NULL, "HP Amp"}, {"HPOL", NULL, "HP L Amp"}, + {"HPOR", NULL, "HP Amp"}, {"HPOR", NULL, "HP R Amp"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, @@ -1792,17 +2083,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +2133,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..cb6bce9 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,10 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool spo_l_mute; + bool spo_r_mute; + bool hp_l_mute; + bool hp_r_mute; }; #endif -- 1.8.1.1.439.g50a6b54

5 26

[alsa-devel] [PATCH] ASoC: compress: Use power efficient workqueue
by Mark Brown 12 Aug '13

12 Aug '13

From: Mark Brown <broonie(a)linaro.org> There is no need for the power down work to be done on a per CPU workqueue especially considering the fairly long delay before powerdown. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d220150..53c9ecd 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -149,8 +149,9 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SND_SOC_DAPM_STREAM_STOP); } else { rtd->pop_wait = 1; - schedule_delayed_work(&rtd->delayed_work, - msecs_to_jiffies(rtd->pmdown_time)); + queue_delayed_work(system_power_efficient_wq, + &rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } } else { /* capture streams can be powered down now */ -- 1.8.4.rc1

2 1

[alsa-devel] BeagleBoard driver development (PCM3168)
by Michael Jordan 12 Aug '13

12 Aug '13

Okay, before I went too far I thought to download archives and found that Klimann has been working on this, but last post I see is April 2013. I picked through and got what I think are the last discussed of the 3 files omap-pcm3168.c pcm3168.c and pcm3168.h. I did not mean to cross paths with anybody. I will compare these to what I have worked up and see what I can contribute. I am a bit confused about one thing though. When I began this (before I knew I might be stepping on somebody else's work) I downloaded 3.10 sources from kernel.org. I have previously and again grepped through the sources and I find no mention of pcm3168. Has this project just not made it that far? And do I have the latest versions (plucked from the text archives, or are they in a git somewhere that I don't know about? Michael

1 0

[alsa-devel] Hello
by Michael Jordan 11 Aug '13

11 Aug '13

Hello all, I am involved in a university project using a TI OMAPL138 EXPKIT (Logic PD Zoom Board) to create a sound processing box. I am tasked with developing the code to replace the TLV320AIC3106 with a PCM3168A due to the greater ADC/DAC count. We have already accomplished the hardware end, and I have been exercising my linux skills at trying to put this together. I found this list and subscribed, and I am glad I sat and listened for a bit before I said anything. 457 patches in the last 10 days! My fingers would fall off! I will get together what I have been working on (2 files pcm3168a.c and pcm3168a.h) and post them. I have some questions, obviously, but I will follow up with them in a bit. If someone could help me along it would be greatly appreciated. Or if I can just provide some relevant information, I would be happy to do that as well. Thanks, Michael

1 0

[alsa-devel] [PATCH] ASoC: doc: Add documentation to MAINTAINERS patterns
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 427a930..07e2bc8 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7710,6 +7710,7 @@ T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported +F: Documentation/sound/alsa/soc/ F: sound/soc/ F: include/sound/soc* -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH 1/16] sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument
by Julia Lawall 11 Aug '13

11 Aug '13

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the arity of the called function. The semantic match that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,f; @@ f(..., - ARRAY_AND_SIZE(e) + e,ARRAY_SIZE(e) ,...) // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not compiled. sound/soc/pxa/mioa701_wm9713.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e..20fdce6 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -133,10 +133,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG);

2 1

[alsa-devel] [PATCH] ASoC: cs4271: fix dapm routing table
by Daniel Mack 11 Aug '13

11 Aug '13

The widgets are names "AINA" and "AINB", which is also what the datasheets calls the terminals. Fix the routing table to match these names. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 35a6a67..a20f1bb 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -184,8 +184,8 @@ SND_SOC_DAPM_OUTPUT("AOUTB-"), }; static const struct snd_soc_dapm_route cs4271_dapm_routes[] = { - { "Capture", NULL, "AINL" }, - { "Capture", NULL, "AINR" }, + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, { "AOUTA+", NULL, "Playback" }, { "AOUTA-", NULL, "Playback" }, -- 1.8.3.1

2 1

[alsa-devel] [PATCH] usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
by Maksim A. Boyko 11 Aug '13

11 Aug '13

Add the volume control quirk for avoiding the kernel warning for the Logitech HD Webcam C525 as in the similar commit 36691e1be6ec551eef4a5225f126a281f8c051c2 (ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310) for the Logitech HD Webcam C310. Reported-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Tested-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 3.10.5+ Signed-off-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> -- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ static void volume_control_quirks(struct case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- WBR, Maksim Boyko

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs4271: Convert to table based control init
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs4271.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 03036b3..65ad56c 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -576,8 +576,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) CS4271_MODE2_MUTECAEQUB, CS4271_MODE2_MUTECAEQUB); - return snd_soc_add_codec_controls(codec, cs4271_snd_controls, - ARRAY_SIZE(cs4271_snd_controls)); + return 0; } static int cs4271_remove(struct snd_soc_codec *codec) @@ -596,6 +595,9 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, + + .controls = cs4271_snd_controls, + .num_controls = ARRAY_SIZE(cs4271_snd_controls), }; #if defined(CONFIG_SPI_MASTER) -- 1.8.4.rc1

3 5

[alsa-devel] [PATCH] ASoC: max9768: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/max9768.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index a6ac231..31f9156 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -118,6 +118,18 @@ static const struct snd_kcontrol_new max9768_mute[] = { SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; +static const struct snd_soc_dapm_widget max9768_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN"), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +}; + +static const struct snd_soc_dapm_route max9768_dapm_routes[] = { + { "OUT+", NULL, "IN" }, + { "OUT-", NULL, "IN" }, +}; + static int max9768_probe(struct snd_soc_codec *codec) { struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); @@ -148,6 +160,10 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .probe = max9768_probe, .controls = max9768_volume, .num_controls = ARRAY_SIZE(max9768_volume), + .dapm_widgets = max9768_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9768_dapm_widgets), + .dapm_routes = max9768_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9768_dapm_routes), }; static const struct regmap_config max9768_i2c_regmap_config = { -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] ASoC: ak5386: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak5386.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 1f30398..72e953b 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -22,7 +22,22 @@ struct ak5386_priv { int reset_gpio; }; -static struct snd_soc_codec_driver soc_codec_ak5386; +static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + +static struct snd_soc_codec_driver soc_codec_ak5386 = { + .dapm_widgets = ak5386_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), + .dapm_routes = ak5386_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak5386_dapm_routes), +}; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] ASoC: ak5386: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak5386.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 1f30398..fb01f35 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -22,7 +22,22 @@ struct ak5386_priv { int reset_gpio; }; -static struct snd_soc_codec_driver soc_codec_ak5386; +static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route ak4554_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + +static struct snd_soc_codec_driver soc_codec_ak5386 = { + .dapm_widgets = ak4554_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), + .dapm_routes = ak4554_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes), +}; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) -- 1.8.4.rc1

2 2

[alsa-devel] [PATCH] ASoC: cs4270: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cs4270.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8e47798..83c835d 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -139,6 +139,22 @@ struct cs4270_private { struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; +static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA", NULL, "Playback" }, + { "AOUTB", NULL, "Playback" }, +}; + /** * struct cs4270_mode_ratios - clock ratio tables * @ratio: the ratio of MCLK to the sample rate @@ -612,6 +628,10 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .controls = cs4270_snd_controls, .num_controls = ARRAY_SIZE(cs4270_snd_controls), + .dapm_widgets = cs4270_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4270_dapm_widgets), + .dapm_routes = cs4270_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4270_dapm_routes), }; /* -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] ASoC: ak4554: Add DAPM support
by Mark Brown 11 Aug '13

11 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4554.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index 6aed9c4..79e9555 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -29,6 +29,22 @@ * CPU-DAI2 (capture only fmt = LEFT_J) ---+ */ +static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route ak4554_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, + + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ak4554_dai = { .name = "ak4554-hifi", .playback = { @@ -49,6 +65,10 @@ static struct snd_soc_dai_driver ak4554_dai = { }; static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .dapm_widgets = ak4554_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), + .dapm_routes = ak4554_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes), }; static int ak4554_soc_probe(struct platform_device *pdev) -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH 1/3] ASoC: bt-sco: enable OF support
by Barry Song 11 Aug '13

11 Aug '13

From: Rongjun Ying <Rongjun.Ying(a)csr.com> this makes users be able to add a bt-sco node in dts instead of hardcoding a platform device in mach. Signed-off-by: Rongjun Ying <Rongjun.Ying(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/bt-sco.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index a081d9f..39e7ebe 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -46,6 +46,14 @@ static int bt_sco_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id bt_sco_of_match[] = { + { .compatible = "bt-sco-codec", }, + {} +}; +MODULE_DEVICE_TABLE(of, bt_sco_of_match); +#endif + static struct platform_device_id bt_sco_driver_ids[] = { { .name = "dfbmcs320", @@ -58,6 +66,9 @@ static struct platform_driver bt_sco_driver = { .driver = { .name = "bt-sco", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = bt_sco_of_match, +#endif }, .probe = bt_sco_probe, .remove = bt_sco_remove, -- 1.8.2.3 Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Follow CSR on Twitter at http://twitter.com/CSR_PLC and read our blog at www.csr.com/blog

5 15

[alsa-devel] [PATCH 1/2] ASoC: Remove unused snd_soc_info_volsw_ext()
by Lars-Peter Clausen 11 Aug '13

11 Aug '13

The SOC_SINGLE_EXT control has been using snd_soc_info_volsw() for its info callback since commit 1c433fb ("[ALSA] soc - 0.13 ASoC headers"). The snd_soc_info_volsw_ext() function has been unused ever since then, so remove it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 -- sound/soc/soc-core.c | 26 -------------------------- 2 files changed, 28 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 13d388a..64eb60a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -499,8 +499,6 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); #define snd_soc_info_bool_ext snd_ctl_boolean_mono_info int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fdddc45..47cb2c0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2594,32 +2594,6 @@ int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); /** - * snd_soc_info_volsw_ext - external single mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a single external mixer control. - * - * Returns 0 for success. - */ -int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int max = kcontrol->private_value; - - if (max == 1 && !strstr(kcontrol->id.name, " Volume")) - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - else - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); - -/** * snd_soc_info_volsw - single mixer info callback * @kcontrol: mixer control * @uinfo: control element information -- 1.8.0

2 2

[alsa-devel] [PATCH v10 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 11 Aug '13

11 Aug '13

Hi, this series implements ac97-slave support for fsl-ssi and phycore ac97 DT support. Since v9 there were a lot of changes. I dropped the board specific ac97-link reset functions and added generic functions to soc-core. It uses devicetree parsed pinctrl and gpios. The audmux setup was removed from phycore-ac97-dt. The initial setup can now be defined in DT. Regards, Markus Markus Pargmann (7): ASoC: core: Generic ac97 link reset functions ASoC: codec: wm9712 simple DT bindings ASoC: imx-audmux: Read default configuration from devicetree ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: ac97-slave support ASoC: Add phycore-ac97-dt driver .../devicetree/bindings/sound/fsl,ssi.txt | 12 + .../devicetree/bindings/sound/imx-audmux.txt | 9 + .../bindings/sound/phytec,phycore-ac97.txt | 16 + .../devicetree/bindings/sound/soc-ac97link.txt | 28 ++ Documentation/devicetree/bindings/sound/wm9712.txt | 10 + include/sound/soc.h | 2 + sound/soc/codecs/wm9712.c | 9 + sound/soc/fsl/Kconfig | 15 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 431 +++++++++++++++++---- sound/soc/fsl/imx-audmux.c | 62 +++ sound/soc/fsl/phycore-ac97-dt.c | 124 ++++++ sound/soc/fsl/phycore-ac97.c | 13 +- sound/soc/soc-core.c | 153 ++++++++ 14 files changed, 789 insertions(+), 97 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 Documentation/devicetree/bindings/sound/wm9712.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

2 19

[alsa-devel] [PATCH] usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
by Maksim A. Boyko 10 Aug '13

10 Aug '13

Add the volume control quirk for avoiding the kernel warning for the Logitech HD Webcam C525 as in the similar commit 36691e1be6ec551eef4a5225f126a281f8c051c2 for the Logitech HD Webcam C310. Reported-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Tested-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 3.10.5+ Signed-off-by: Maksim Boyko <maksim.a.boyko(a)gmail.com> -- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ static void volume_control_quirks(struct case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- WBR, Maksim P.S. Please, use this patch due to it should have correct tab symbols.

3 2

[alsa-devel] [PATCH] ALSA: Add MAINTAINERS entry for compressed audio API
by Mark Brown 10 Aug '13

10 Aug '13

From: Mark Brown <broonie(a)linaro.org> Help ensure that Vinod gets included in review of compressed audio patches by adding a MAINTAINERS entry for it. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- MAINTAINERS | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index a367e52..427a930 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7694,6 +7694,15 @@ F: include/sound/ F: include/uapi/sound/ F: sound/ +SOUND - COMPRESSED AUDIO +M: Vinod Koul <vinod.koul(a)intel.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git +S: Supported +F: include/sound/compress_driver.h +F: sound/core/compress_offload.c +F: sound/soc/soc-compress.c + SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> -- 1.8.4.rc1

2 1

[alsa-devel] [PATCH] ASoC: Remove unused soc_pm_waitq
by Lars-Peter Clausen 10 Aug '13

10 Aug '13

The soc_pm_waitq waitqueue has been around as long as the ASoC framework existed, but has never been used so far, so remove it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1d5e84a..fdddc45 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -47,8 +47,6 @@ #define NAME_SIZE 32 -static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); - #ifdef CONFIG_DEBUG_FS struct dentry *snd_soc_debugfs_root; EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); -- 1.8.0

2 1

[alsa-devel] [PATCH v4 1/1] ASoc: kirkwood: add DT support to the mvebu audio subsystem
by Jean-Francois Moine 10 Aug '13

10 Aug '13

This patch adds DT support to the audio subsystem of the mvebu family (Kirkwood, Dove, Armada 370). Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/mvebu-audio.txt | 29 ++++++++++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.c | 26 +++++++++++++++++----- 2 files changed, 49 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 0000000..7e5fd37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) audio controller + +Required properties: + +- compatible: "mrvl,mvebu-audio" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "extclk" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "mrvl,mvebu-audio"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c.next b/sound/soc/kirkwood/kirkwood-i2s.c index e5f3f7a..a4170b4 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c.next +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -22,6 +22,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/platform_data/asoc-kirkwood.h> +#include <linux/of.h> + #include "kirkwood.h" #define DRV_NAME "mvebu-audio" @@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); return -EINVAL; } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); return PTR_ERR(priv->clk); @@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id kirkwood_i2s_of_match[] = { + { .compatible = "mrvl,mvebu-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, kirkwood_i2s_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(kirkwood_i2s_of_match), }, }; -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

6 20

[alsa-devel] [patch] sound: oss/dmabuf: remove an unneeded temporary variable
by Dan Carpenter 09 Aug '13

09 Aug '13

We don't actually use the "go" variable so it can be removed. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index a59c888..461d94c 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) unsigned long flags; int err = 0, n = 0; struct dma_buffparms *dmap = adev->dmap_in; - int go; if (!(adev->open_mode & OPEN_READ)) return -EIO; @@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) spin_unlock_irqrestore(&dmap->lock,flags); return -EAGAIN; } - if ((go = adev->go)) + if (adev->go) timeout = dmabuf_timeout(dmap); spin_unlock_irqrestore(&dmap->lock,flags);

2 2

[alsa-devel] [PATCH V3 0/3] Add audio support on smdk5420
by Padmavathi Venna 09 Aug '13

09 Aug '13

Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control, internal mux for rclk src clk and tdm support. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. This also adds i2s support on exynos5420. Changes since V2: - Seperated out driver side changes and dts changes in two patch sets. - Replaced samsung,s3c6410-i2s-v4 with samsung,s3c6410-i2s-multi for more clarity as suggested by Tomasz Figa. Changes since V1: - Pass quirks as driver data and fetch the quirks based on compatible string from dtsi file as suggested by Tomasz Figa and Mark Brown - Make the I2S driver more flexible with respect to register access as suggested by Tomasz Figa and Mark Brown - Add 5420 support in the driver. - Modify the dtsi files with the corresponding compatible strings and removed the i2s quirks from 5250 dtsi file. - Updated the i2s Documentation with relevent changes and i2s versioning info. - Add i2s nodes on exynos5420.dtsi - Enable sound support on smdk5420 This patch set is made based on Mark Brown for-next branch on sound.git. Padmavathi Venna (3): platform: Increase platform name size ASoC: Samsung: I2S: Add quirks as driver data in I2S ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 .../devicetree/bindings/sound/samsung-i2s.txt | 25 ++-- include/linux/mod_devicetable.h | 2 +- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 15 ++ sound/soc/samsung/i2s.c | 163 +++++++++++++++----- 5 files changed, 158 insertions(+), 48 deletions(-) -- 1.7.4.4

3 5

[alsa-devel] Found a typo
by Torino 09 Aug '13

09 Aug '13

Dear Alsa developers, within Saucy (13.10), "alsa-utils", string nr. 124 it says "Channel %d doesn't match with hw_parmas" whereas in other strings like nr. 84 or 126 it says "hw_params". I hope it was correct informing you via this mailing list. Best Regads, Torino (from the German Übersetzer-Team)

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: do not trust too-big wMaxPacketSize values
by Clemens Ladisch 08 Aug '13

08 Aug '13

The driver used to assume that the streaming endpoint's wMaxPacketSize value would be an indication of how much data the endpoint expects or sends, and compute the number of packets per URB using this value. However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes, while only about 88 or 44 bytes are be actually used. This discrepancy would result in URBs with far too few packets, which would not work correctly on the EHCI driver. To get correct URBs, use wMaxPacketSize only as an upper limit on the packet size. Reported-by: James Stone <jamesmstone(a)gmail.com> Tested-by: James Stone <jamesmstone(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 2.6.35+ Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/endpoint.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -591,17 +591,16 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate max. frequency */ - if (ep->maxpacksize) { + /* assume max. frequency is 25% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ maxsize = ep->maxpacksize; ep->freqmax = (maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); } if (ep->fill_max)

3 2

[alsa-devel] [PATCH 1/1] ASoC: s6105-ipcam: Fix incorrect placement of __initdata
by Sachin Kamat 08 Aug '13

08 Aug '13

__initdata should be placed between the variable name and equal sign for the variable to be placed in the intended section. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> Cc: Daniel Gloeckner <dg(a)emlix.com> --- sound/soc/s6000/s6105-ipcam.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 58cfb1e..945e8ab 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -192,7 +192,7 @@ static struct snd_soc_card snd_soc_card_s6105 = { .num_links = 1, }; -static struct s6000_snd_platform_data __initdata s6105_snd_data = { +static struct s6000_snd_platform_data s6105_snd_data __initdata = { .wide = 0, .channel_in = 0, .channel_out = 1, -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] ASoC: ad73311: Add DAPM support
by Mark Brown 08 Aug '13

08 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it possible to hook up other devices in boards and is required by removal of support for non-DAPM CODECs in the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad73311.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b1f2baf..5fac8ad 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -23,6 +23,21 @@ #include "ad73311.h" +static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINP"), +SND_SOC_DAPM_INPUT("VINN"), +SND_SOC_DAPM_OUTPUT("VOUTN"), +SND_SOC_DAPM_OUTPUT("VOUTP"), +}; + +static const struct snd_soc_dapm_route ad73311_dapm_routes[] = { + { "Capture", NULL, "VINP" }, + { "Capture", NULL, "VINN" }, + + { "VOUTN", NULL, "Playback" }, + { "VOUTP", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ad73311_dai = { .name = "ad73311-hifi", .playback = { @@ -39,7 +54,12 @@ static struct snd_soc_dai_driver ad73311_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -static struct snd_soc_codec_driver soc_codec_dev_ad73311; +static struct snd_soc_codec_driver soc_codec_dev_ad73311 = { + .dapm_widgets = ad73311_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets), + .dapm_routes = ad73311_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes), +}; static int ad73311_probe(struct platform_device *pdev) { -- 1.8.4.rc1

2 3

[alsa-devel] [PATCH] ASoC: pcm1681: Staticise DAI driver
by Mark Brown 08 Aug '13

08 Aug '13

From: Mark Brown <broonie(a)linaro.org> It is not exported so doesn't need to be in the global namespace and sparse warns on this. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 27da41b..51b1866 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -225,7 +225,7 @@ static const struct snd_kcontrol_new pcm1681_controls[] = { pcm1681_get_deemph, pcm1681_put_deemph), }; -struct snd_soc_dai_driver pcm1681_dai = { +static struct snd_soc_dai_driver pcm1681_dai = { .name = "pcm1681-hifi", .playback = { .stream_name = "Playback", -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH v4 0/1] ASoc: kirkwood: extend the kirkwood audio subsystem for DT usage
by Jean-Francois Moine 08 Aug '13

08 Aug '13

This patch series adds DT support to the kirkwood/mvebu audio subsystem. History v4 - change the DT name to "mrvl,mvebu-audio" - remove previous patches 1..3 which have been applied by Russell King v3 - change the DT name from "kirkwood-i2s" to "mvebu-pcm-audio" - apply to linux-next - let NULL as the internal i2s clock for compatibility - change i2s external clock name to "extclk" instead of "external" - cancel previous patch 1/4. Russell's patch should be applied instead. v2 - fix compilation errors -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

1 0

[alsa-devel] bug
by Mattia Bertoni 08 Aug '13

08 Aug '13

finally I can report a bug which I encountered some time ago; I send you the important part of the entire log, which is to much big to be posted: Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 420736 bytes (2191 ms). Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_dump(): Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Hardware PCM card 0 'NVidia CK804' device 0 subdevice 0 Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Its setup is: Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stream : PLAYBACK Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: access : MMAP_INTERLEAVED Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: format : S16_LE Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: subformat : STD Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: channels : 2 Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: rate : 48000 Jun 15 18:18:00 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: exact rate : 48000 (48000/1) Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: msbits : 16 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: buffer_size : 16384 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_size : 16384 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_time : 341333 Jun 15 18:18:03 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: tstamp_mode : ENABLE Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_step : 1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: avail_min : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_event : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: start_threshold : -1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stop_threshold : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_threshold: 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_size : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: boundary : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: appl_ptr : 26406104 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: hw_ptr : 26494904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -465056 bytes (-2422 ms). Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: snd_pcm_dump(): Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Hardware PCM card 0 'NVidia CK804' device 0 subdevice 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: Its setup is: Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stream : PLAYBACK Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: access : MMAP_INTERLEAVED Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: format : S16_LE Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: subformat : STD Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: channels : 2 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: rate : 48000 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: exact rate : 48000 (48000/1) Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: msbits : 16 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: buffer_size : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_size : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_time : 341333 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: tstamp_mode : ENABLE Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_step : 1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: avail_min : 16384 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: period_event : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: start_threshold : -1 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: stop_threshold : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_threshold: 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: silence_size : 0 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: boundary : 4611686018427387904 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: appl_ptr : 26569944 Jun 15 18:18:04 arch-dekstop pulseaudio[776]: [alsa-sink-Intel ICH] alsa-util.c: hw_ptr : 26686208

2 6

[alsa-devel] [PATCH] [alsa-lib] UCM: Document some standard values
by Arun Raghavan 08 Aug '13

08 Aug '13

Playback/CaptureChannels has been around for a bit. Playback/CaptureRate is new and is intended to be used to specify the sample rate at which the Playback/CapturePCM should be opened. --- include/use-case.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/use-case.h b/include/use-case.h index 93b3456..4e13249 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -259,9 +259,13 @@ int snd_use_case_get_list(snd_use_case_mgr_t *uc_mgr, * TQ - Tone Quality * PlaybackPCM - full PCM playback device name * CapturePCM - full PCM capture device name + * PlaybackRate - playback device sample rate + * PlaybackChannels - playback device channel count * PlaybackCTL - playback control device name * PlaybackVolume - playback control volume ID string * PlaybackSwitch - playback control switch ID string + * CaptureRate - capture device sample rate + * CaptureChannels - capture device channel count * CaptureCTL - capture control device name * CaptureVolume - capture control volume ID string * CaptureSwitch - capture control switch ID string -- 1.8.2.1

2 1

Re: [alsa-devel] [PATCH V3 2/3] ASoC: Samsung: I2S: Add quirks as driver data in I2S
by Padma Venkat 08 Aug '13

08 Aug '13

Hi Tomasz, On Wed, Aug 7, 2013 at 4:43 PM, Tomasz Figa <t.figa(a)samsung.com> wrote: > Ahh, one more thing inline. > [snip] >> >> +static struct samsung_i2s_dai_data i2sv3_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv4_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_c100_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR | QUIRK_SEC_DAI | >> + QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_dai_type = { >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data samsung_dai_type_sec = { >> + .dai_type = TYPE_SEC, >> +}; >> + >> static struct platform_device_id samsung_i2s_driver_ids[] = { >> { >> - .name = "samsung-i2s", >> - .driver_data = TYPE_PRI, >> + .name = "samsung,s3c6410-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, >> + }, { >> + .name = "samsung,s3c6410-i2s-multi", >> + .driver_data = (kernel_ulong_t)&i2sv4_dai_type, >> + }, { >> + .name = "samsung,s5pc100-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_c100_dai_type, >> }, { >> - .name = "samsung-i2s-sec", >> - .driver_data = TYPE_SEC, >> + .name = "samsung,s5pv210-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_dai_type, >> + }, { >> + .name = "samsung-i2s-sec", >> + .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, > > I don't think you need to change the legacy platform IDs at all. If legacy platforms not required to change then I need to introduce a new samsung_i2s_dai_data structure which holds only dai_type for non-dt platforms. If I change legacy platforms it breaks the older platforms now. Is it okay adding a samsung_i2s_dai_data structure for non-dt platforms which will be removed later? > > IMHO it would be better to keep the old way of quirk retrieval from > platform_data (which I think this patch does anyway, because the probe path > without DT is unchanged), as it will be dropped in some point in time > anyway. > > This would also eliminate the need for patch 1. > > Best regards, > Tomasz > Thanks Padma

3 2

[alsa-devel] snd_pcm_set_params typo patch
by Laurentiu Nicola 08 Aug '13

08 Aug '13

Please, somebody apply this one. I lost quite a bit of time because of it :). Also, is there anything wrong with the bug tracker? It doesn't load for me. diff -urpN alsa-lib-1.0.27.2/src/pcm/pcm.c alsa-lib-1.0.27.2-mine/src/pcm/pcm.c --- alsa-lib-1.0.27.2/src/pcm/pcm.c 2013-07-08 15:31:36.000000000 +0300 +++ alsa-lib-1.0.27.2-mine/src/pcm/pcm.c 2013-08-07 22:17:14.908673700 +0300 @@ -7932,7 +7932,7 @@ int snd_pcm_set_params(snd_pcm_t *pcm, return err; } if (rrate != rate) { - SNDERR("Rate doesn't match (requested %iHz, get %iHz)", rate, err); + SNDERR("Rate doesn't match (requested %iHz, got %iHz)", rate, rrate); return -EINVAL; } /* set the buffer time */ -- Because there are many ways to skin a cat, but you only get to choose one.

2 1

[alsa-devel] [PATCH] ALSA: don't push static constants on stack for %*ph
by Andy Shevchenko 08 Aug '13

08 Aug '13

There is no need to pass constants via stack. The width may be explicitly specified in the format. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/isa/gus/interwave.c | 3 +-- sound/usb/6fire/firmware.c | 4 ++-- 2 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 9942691..afef0d7 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); #ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos, - 8, iwave); + printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave); #endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index b9defcd..780bf3f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version) if (!memcmp(version, known_fw_versions + i, 2)) return 0; - snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. " "please reconnect to power. if this failure " "still happens, check your firmware installation.", - 4, version); + version); return -EINVAL; } -- 1.8.4.rc1

2 1

[alsa-devel] Alsa bug/bugtracker
by Mattia Bertoni 08 Aug '13

08 Aug '13

hi, under Mark Brown directions I submit to you my problem: recently I've gotten a crash on my machine running Archlinux x86_64 the log suggest me to report it in the Alsa bugtracker but the link on the official project (https://bugtrack.alsa-project.org/alsa-bug/) is down.... how can I do ??? thank you

3 4

[alsa-devel] [PATCH] Fix invalid volume resolution for Logitech HD Webcam C525
by Бойко Максим 08 Aug '13

08 Aug '13

From: Maksim Boyko <maksim.a.boyko(a)gmail.com> Add the volume control quirk for avoiding the kernel warning for Logitech HD Webcam C525. The similar patch was previously reported for Logitech HD Webcam C310 (see 36691e1be6ec551eef4a5225f126a281f8c051c2). --- linux-3.10.5-orig/sound/usb/mixer.c 2013-08-04 12:51:49.000000000 +0400 +++ linux-3.10.5/sound/usb/mixer.c 2013-08-06 21:12:42.274197287 +0400 @@ -888,6 +888,7 @@ case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384.

3 2

[alsa-devel] [PATCH] ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding
by Mark Brown 08 Aug '13

08 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 24 ++-------------------- Documentation/devicetree/bindings/sound/wm8903.txt | 19 +++++++++++++++++ 2 files changed, 21 insertions(+), 22 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt index 3bf722d..4b44dfb 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt @@ -11,28 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins, and the jacks on the board: - - WM8903 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS - - Board connectors: + sinks are the WM8903's pins (documented in the WM8903 binding document), + and the jacks on the board: * Headphone Jack * Int Spk diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index f102cbc..94ec32c 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -28,6 +28,25 @@ Optional properties: performed. If any entry has the value 0xffffffff, that GPIO's configuration will not be modified. +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + Example: codec: wm8903@1a { -- 1.8.4.rc1

3 2

[alsa-devel] [PATCH] ALSA: hda - Fix missing mute controls for CX5051
by Takashi Iwai 08 Aug '13

08 Aug '13

We've added a fake mute control (setting the amp volume to zero) for CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but this feature was overlooked in the generic parser implementation. Now the driver lacks of mute controls on these codecs. The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE bits in each place checking the amp capabilities. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001 Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbb..e3c7ba8 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, } #define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE) + check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) #define nid_has_volume(codec, nid, dir) \ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) @@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, if (enable) val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; } - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (!enable) val |= HDA_AMP_MUTE; } @@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, { unsigned int mask = 0xff; - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL)) mask &= ~0x80; } -- 1.8.3.1

1 0

[alsa-devel] [PATCH 1/2] ASoC: ak4104: Add stub DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it easer to integrate the device with other on-board components and ensures correct operation following removal of support for non-DAPM CODECs. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4104.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c7cfdf9..9a7c89b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -51,6 +51,14 @@ struct ak4104_private { struct regmap *regmap; }; +static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { + { "TX", NULL, "Playback" }, +}; + static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { @@ -214,6 +222,11 @@ static int ak4104_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, + + .dapm_widgets = ak4104_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets), + .dapm_routes = ak4104_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes), }; static const struct regmap_config ak4104_regmap = { -- 1.8.4.rc1

1 1

[alsa-devel] [PATCH] ASoC: dt: Move WM8903 pin list from Tegra board binding to CODEC binding
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> The pin list is the same for any board using the CODEC. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- .../bindings/sound/nvidia,tegra-audio-wm8903.txt | 22 ++-------------------- Documentation/devicetree/bindings/sound/wm8903.txt | 19 +++++++++++++++++++ 2 files changed, 21 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt index 3bf722d..0efe58c 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt @@ -11,26 +11,8 @@ Required properties: - nvidia,audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins, and the jacks on the board: - - WM8903 pins: - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS + sinks are the WM8903's pins (documented in the WM8903 binding document), + and the jacks on the board: Board connectors: diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt index f102cbc..94ec32c 100644 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ b/Documentation/devicetree/bindings/sound/wm8903.txt @@ -28,6 +28,25 @@ Optional properties: performed. If any entry has the value 0xffffffff, that GPIO's configuration will not be modified. +Pins on the device (for linking into audio routes): + + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + Example: codec: wm8903@1a { -- 1.8.4.rc1

2 4

[alsa-devel] [PATCH] ASoC: ads711x: Add DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> This makes it easier to hook into boards and ensures the driver continues to work with support for non-DAPM CODECs removed. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ads117x.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 506d474..8f388ed 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -23,6 +23,28 @@ #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) +static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("Input1"), +SND_SOC_DAPM_INPUT("Input2"), +SND_SOC_DAPM_INPUT("Input3"), +SND_SOC_DAPM_INPUT("Input4"), +SND_SOC_DAPM_INPUT("Input5"), +SND_SOC_DAPM_INPUT("Input6"), +SND_SOC_DAPM_INPUT("Input7"), +SND_SOC_DAPM_INPUT("Input8"), +}; + +static const struct snd_soc_dapm_route ads117x_dapm_routes[] = { + { "Capture", NULL, "Input1" }, + { "Capture", NULL, "Input2" }, + { "Capture", NULL, "Input3" }, + { "Capture", NULL, "Input4" }, + { "Capture", NULL, "Input5" }, + { "Capture", NULL, "Input6" }, + { "Capture", NULL, "Input7" }, + { "Capture", NULL, "Input8" }, +}; + static struct snd_soc_dai_driver ads117x_dai = { /* ADC */ .name = "ads117x-hifi", @@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = { .formats = ADS117X_FORMATS,}, }; -static struct snd_soc_codec_driver soc_codec_dev_ads117x; +static struct snd_soc_codec_driver soc_codec_dev_ads117x = { + .dapm_widgets = ads117x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets), + .dapm_routes = ads117x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes), +}; static int ads117x_probe(struct platform_device *pdev) { -- 1.8.4.rc1

1 0

[alsa-devel] [PATCH] 6fire: fix DMA issues with URB transfer_buffer usage
by Jussi Kivilinna 07 Aug '13

07 Aug '13

Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to be DMA-able, which stack is not. Furthermore, transfer_buffer should not be allocated as part of larger device structure because DMA coherency issues and patch fixes this issue too. Patch is only compile tested. Cc: stable(a)vger.kernel.org Signed-off-by: Jussi Kivilinna <jussi.kivilinna(a)iki.fi> --- sound/usb/6fire/comm.c | 38 +++++++++++++++++++++++++++++++++----- sound/usb/6fire/comm.h | 2 +- 2 files changed, 34 insertions(+), 6 deletions(-) diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 9e6e3ff..23452ee 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -110,19 +110,37 @@ static int usb6fire_comm_send_buffer(u8 *buffer, struct usb_device *dev) static int usb6fire_comm_write8(struct comm_runtime *rt, u8 request, u8 reg, u8 value) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, value, 0x00); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } static int usb6fire_comm_write16(struct comm_runtime *rt, u8 request, u8 reg, u8 vl, u8 vh) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, vl, vh); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } int usb6fire_comm_init(struct sfire_chip *chip) @@ -135,6 +153,12 @@ int usb6fire_comm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->receiver_buffer = kzalloc(COMM_RECEIVER_BUFSIZE, GFP_KERNEL); + if (!rt->receiver_buffer) { + kfree(rt); + return -ENOMEM; + } + urb = &rt->receiver; rt->serial = 1; rt->chip = chip; @@ -153,6 +177,7 @@ int usb6fire_comm_init(struct sfire_chip *chip) urb->interval = 1; ret = usb_submit_urb(urb, GFP_KERNEL); if (ret < 0) { + kfree(rt->receiver_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create comm data receiver."); return ret; @@ -171,6 +196,9 @@ void usb6fire_comm_abort(struct sfire_chip *chip) void usb6fire_comm_destroy(struct sfire_chip *chip) { - kfree(chip->comm); + struct comm_runtime *rt = chip->comm; + + kfree(rt->receiver_buffer); + kfree(rt); chip->comm = NULL; } diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 6a0840b..780d5ed 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -24,7 +24,7 @@ struct comm_runtime { struct sfire_chip *chip; struct urb receiver; - u8 receiver_buffer[COMM_RECEIVER_BUFSIZE]; + u8 *receiver_buffer; u8 serial; /* urb serial */

3 3

Re: [alsa-devel] [PATCH V3 2/3] ASoC: Samsung: I2S: Add quirks as driver data in I2S
by Padma Venkat 07 Aug '13

07 Aug '13

Hi Tomasz, On Wed, Aug 7, 2013 at 4:32 PM, Tomasz Figa <t.figa(a)samsung.com> wrote: > Hi Padmavathi, > > [Ccing DT maintainers with a little comment about contents of this patch > for them: > > This is a rework of Samsung i2s bindings to make them more reasonable than > they are at the moment. I know this was supposed to be stable, fixed, ABI, > etc., but as we discussed a lot the whole topic of DT bindings, we need to > sanitize existing bindings and this patch is a part of this work.] > [snip] >> >> Required SoC Specific Properties: >> >> -- compatible : "samsung,i2s-v5" >> +- compatible : should be one of the following. >> + - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. Previous versions >> + has only 8/16bit support. > > The comment about previous versions seems a bit enigmatic. > > If there is another variant supported by this driver that supports only > 8/16bit audio data, then it should have a separate compatible value. For previous i2s controllers there are two other i2s drivers available. I think I can delete this comment here. > >> + - samsung,s3c6410-i2sv4: for 8/16/24bit multichannel(5.1 channel) >> I2S. + Introduced in s3c6410. This also applicable for s5p64x0 >> platforms. + - samsung,s5pc100-i2s: for 8/16/24bit multichannel(5.1 >> channel) I2S + with secondary fifo and s/w reset control. >> + - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with >> + secondary fifo, s/w reset control and internal mux for root clk >> src. + >> - reg: physical base address of the controller and length of memory >> mapped region. >> - dmas: list of DMA controller phandle and DMA request line ordered >> pairs. @@ -21,13 +30,6 @@ Required SoC Specific Properties: >> >> Optional SoC Specific Properties: >> >> -- samsung,supports-6ch: If the I2S Primary sound source has 5.1 Channel >> - support, this flag is enabled. >> -- samsung,supports-rstclr: This flag should be set if I2S software reset >> bit - control is required. When this flag is set I2S software reset bit >> will be - enabled or disabled based on need. >> -- samsung,supports-secdai:If I2S block has a secondary FIFO and internal >> DMA, - then this flag is enabled. >> - samsung,idma-addr: Internal DMA register base address of the audio >> sub system(used in secondary sound source). >> - pinctrl-0: Should specify pin control groups used for this controller. >> @@ -46,9 +48,6 @@ i2s0: i2s@03830000 { > > The example has also a compatible value set to "samsung,i2s-v5". I don't > think this is compliant to the new bindings. Yes. It's my mistake. Will change this to new one. > >> <&clock_audss EXYNOS_I2S_BUS>, >> <&clock_audss EXYNOS_SCLK_I2S>; >> clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; >> - samsung,supports-6ch; >> - samsung,supports-rstclr; >> - samsung,supports-secdai; >> samsung,idma-addr = <0x03000000>; >> pinctrl-names = "default"; >> pinctrl-0 = <&i2s0_bus>; >> diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c >> index 47e08dd..8a5504c 100644 >> --- a/sound/soc/samsung/i2s.c >> +++ b/sound/soc/samsung/i2s.c >> @@ -40,6 +40,7 @@ enum samsung_dai_type { >> >> struct samsung_i2s_dai_data { > > const struct samsung_i2s_dai_data OK. > >> int dai_type; >> + u32 quirks; >> }; >> >> struct i2s_dai { >> @@ -1032,18 +1033,18 @@ static struct i2s_dai *i2s_alloc_dai(struct >> platform_device *pdev, bool sec) >> >> static const struct of_device_id exynos_i2s_match[]; >> >> -static inline int samsung_i2s_get_driver_data(struct platform_device >> *pdev) +static inline struct samsung_i2s_dai_data >> *samsung_i2s_get_driver_data( + struct platform_device > *pdev) >> { >> #ifdef CONFIG_OF >> - struct samsung_i2s_dai_data *data; >> if (pdev->dev.of_node) { >> const struct of_device_id *match; >> match = of_match_node(exynos_i2s_match, pdev->dev.of_node); >> - data = (struct samsung_i2s_dai_data *) match->data; >> - return data->dai_type; >> + return (struct samsung_i2s_dai_data *) match->data; > > If you make the dai_data const this cast will be no longer necessary. > >> } else >> #endif >> - return platform_get_device_id(pdev)->driver_data; >> + return (struct samsung_i2s_dai_data *) >> + platform_get_device_id(pdev)->driver_data; >> } >> >> #ifdef CONFIG_PM_RUNTIME >> @@ -1074,13 +1075,13 @@ static int samsung_i2s_probe(struct >> platform_device *pdev) struct resource *res; >> u32 regs_base, quirks = 0, idma_addr = 0; >> struct device_node *np = pdev->dev.of_node; >> - enum samsung_dai_type samsung_dai_type; >> + struct samsung_i2s_dai_data *i2s_dai_data; > > const OK. > >> int ret = 0; >> >> /* Call during Seconday interface registration */ >> - samsung_dai_type = samsung_i2s_get_driver_data(pdev); >> + i2s_dai_data = samsung_i2s_get_driver_data(pdev); >> >> - if (samsung_dai_type == TYPE_SEC) { >> + if (i2s_dai_data->dai_type == TYPE_SEC) { >> sec_dai = dev_get_drvdata(&pdev->dev); >> if (!sec_dai) { >> dev_err(&pdev->dev, "Unable to get drvdata\n"); >> @@ -1129,15 +1130,7 @@ static int samsung_i2s_probe(struct >> platform_device *pdev) idma_addr = i2s_cfg->idma_addr; >> } >> } else { >> - if (of_find_property(np, "samsung,supports-6ch", NULL)) >> - quirks |= QUIRK_PRI_6CHAN; >> - >> - if (of_find_property(np, "samsung,supports-secdai", NULL)) >> - quirks |= QUIRK_SEC_DAI; >> - >> - if (of_find_property(np, "samsung,supports-rstclr", NULL)) >> - quirks |= QUIRK_NEED_RSTCLR; >> - >> + quirks = i2s_dai_data->quirks; >> if (of_property_read_u32(np, "samsung,idma-addr", >> &idma_addr)) { >> if (quirks & QUIRK_SEC_DAI) { >> @@ -1250,27 +1243,60 @@ static int samsung_i2s_remove(struct >> platform_device *pdev) return 0; >> } >> >> +static struct samsung_i2s_dai_data i2sv3_dai_type = { > > const OK. > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv4_dai_type = { > > ditto > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_c100_dai_type = { > > ditto > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_NO_MUXPSR | QUIRK_SEC_DAI | >> + QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data i2sv5_dai_type = { > > ditto > >> + .dai_type = TYPE_PRI, >> + .quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR, >> +}; >> + >> +static struct samsung_i2s_dai_data samsung_dai_type_sec = { > > ditto > >> + .dai_type = TYPE_SEC, >> +}; >> + >> static struct platform_device_id samsung_i2s_driver_ids[] = { >> { >> - .name = "samsung-i2s", >> - .driver_data = TYPE_PRI, >> + .name = "samsung,s3c6410-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv3_dai_type, >> + }, { >> + .name = "samsung,s3c6410-i2s-multi", >> + .driver_data = (kernel_ulong_t)&i2sv4_dai_type, >> + }, { >> + .name = "samsung,s5pc100-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_c100_dai_type, >> }, { >> - .name = "samsung-i2s-sec", >> - .driver_data = TYPE_SEC, >> + .name = "samsung,s5pv210-i2s", >> + .driver_data = (kernel_ulong_t)&i2sv5_dai_type, >> + }, { >> + .name = "samsung-i2s-sec", >> + .driver_data = (kernel_ulong_t)&samsung_dai_type_sec, >> }, >> {}, >> }; >> MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); >> >> #ifdef CONFIG_OF >> -static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { >> - [TYPE_PRI] = { TYPE_PRI }, >> - [TYPE_SEC] = { TYPE_SEC }, >> -}; >> - >> static const struct of_device_id exynos_i2s_match[] = { >> - { .compatible = "samsung,i2s-v5", >> - .data = &samsung_i2s_dai_data_array[TYPE_PRI], >> + { >> + .compatible = "samsung,s3c6410-i2s", >> + .data = &i2sv3_dai_type, >> + }, { >> + .compatible = "samsung,s5pv210-i2s", >> + .data = &i2sv5_dai_type, > > Hmm, documentation mentions 4 compatible values (and possibly one is > missing there), but this patch adds only 2 of them. I don't think this is > correct, especially that the driver already supports handling all of them. Documentation mentions about all the compatible strings which are all supported by this driver for reference. But some of the platforms haven't converted to DT yet so here I added the compatible strings for DT compliant platforms and compatible strings for non-DT compliant platforms I added in the platform_device_id structure. Any way we have to make all legacy platforms DT compliant so we need this "samsung,s3c6410-i2s-multi" in future. > > Best regards, > Tomasz > Thanks Padma

1 0

[alsa-devel] [PATCH] switch: arizona: Clear trig_sts bits on all paths
by Charles Keepax 07 Aug '13

07 Aug '13

We want the trig_sts bits to be cleared in all cases where we consider the jack detection interrupt to have been handled. Specifically, if a duplicate detection event was suppressed these bits were not cleared causing the CODEC to not enter a low power state. This patch clears the bits on the duplicate detection code path. Reported-by: Ryo Tsutsui <ryo.tsutsui(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/extcon/extcon-arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/extcon/extcon-arizona.c b/drivers/extcon/extcon-arizona.c index 7a1b4a7..4df68de 100644 --- a/drivers/extcon/extcon-arizona.c +++ b/drivers/extcon/extcon-arizona.c @@ -1034,6 +1034,7 @@ static irqreturn_t arizona_jackdet(int irq, void *data) else info->micd_timeout = DEFAULT_MICD_TIMEOUT; +out: /* Clear trig_sts to make sure DCVDD is not forced up */ regmap_write(arizona->regmap, ARIZONA_AOD_WKUP_AND_TRIG, ARIZONA_MICD_CLAMP_FALL_TRIG_STS | @@ -1041,7 +1042,6 @@ static irqreturn_t arizona_jackdet(int irq, void *data) ARIZONA_JD1_FALL_TRIG_STS | ARIZONA_JD1_RISE_TRIG_STS); -out: mutex_unlock(&info->lock); pm_runtime_mark_last_busy(info->dev); -- 1.7.2.5

2 2

[alsa-devel] Linux 3.10.4 and Logitech HD Webcam C525/C310
by Бойко Максим 07 Aug '13

07 Aug '13

Hi, I have Logitech HD Webcam 525 (046d:0826) but its mic works strange. The mic is turned off while video works fine. In dmesg I see the following messages [ 472.079738] cannot submit urb (err = -18) [ 472.079748] cannot submit urb (err = -18) [ 472.079754] cannot submit urb (err = -18) [ 472.079759] cannot submit urb (err = -18) [ 472.079783] cannot submit urb (err = -18) [ 472.079787] cannot submit urb (err = -18) [ 472.079792] cannot submit urb (err = -18) [ 472.079797] cannot submit urb (err = -18) I have did the following modifications to try to fix it (it works for some other cams) add to the sound/usb/mixer.c case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ add to the sound/usb/quirks-table.h (see http://forums.gentoo.org/viewtopic-p-6886404.html) { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .idVendor = 0x046d, .idProduct = 0x0826, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Logitech, Inc.", .product_name = "HD Webcam C525", .ifnum = QUIRK_NO_INTERFACE } }, But all these don't fix the mic. I have not enough experience to fix it myself but I can help with finding a bug and testing fixes. I have also found that the same problem exists for Logitech Webcam C310 (046d:081b). The bug is not specific for my laptop. I have the same issue on my desktop with both cameras. See attachments for additional info.

2 3

[alsa-devel] [PATCH 1/4] mfd: wm5110: Mark register containing FLL gains as readable
by Charles Keepax 07 Aug '13

07 Aug '13

The FLL gains are present on all devices and controlled from common code shared between them. This patch adds these registers into defaults list for wm5110 and marks them as readable. Reported-by: D.J. Barrow <dbarrow(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5110-tables.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index 2a79723..c04324d 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -468,12 +468,14 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000176, 0x0000 }, /* R374 - FLL1 Control 6 */ { 0x00000177, 0x0281 }, /* R375 - FLL1 Loop Filter Test 1 */ { 0x00000178, 0x0000 }, /* R376 - FLL1 NCO Test 0 */ + { 0x00000179, 0x0000 }, /* R376 - FLL1 Control 7 */ { 0x00000181, 0x0000 }, /* R385 - FLL1 Synchroniser 1 */ { 0x00000182, 0x0000 }, /* R386 - FLL1 Synchroniser 2 */ { 0x00000183, 0x0000 }, /* R387 - FLL1 Synchroniser 3 */ { 0x00000184, 0x0000 }, /* R388 - FLL1 Synchroniser 4 */ { 0x00000185, 0x0000 }, /* R389 - FLL1 Synchroniser 5 */ { 0x00000186, 0x0000 }, /* R390 - FLL1 Synchroniser 6 */ + { 0x00000187, 0x0001 }, /* R390 - FLL1 Synchroniser 7 */ { 0x00000189, 0x0000 }, /* R393 - FLL1 Spread Spectrum */ { 0x0000018A, 0x0004 }, /* R394 - FLL1 GPIO Clock */ { 0x00000191, 0x0000 }, /* R401 - FLL2 Control 1 */ @@ -484,12 +486,14 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000196, 0x0000 }, /* R406 - FLL2 Control 6 */ { 0x00000197, 0x0000 }, /* R407 - FLL2 Loop Filter Test 1 */ { 0x00000198, 0x0000 }, /* R408 - FLL2 NCO Test 0 */ + { 0x00000199, 0x0000 }, /* R408 - FLL2 Control 7 */ { 0x000001A1, 0x0000 }, /* R417 - FLL2 Synchroniser 1 */ { 0x000001A2, 0x0000 }, /* R418 - FLL2 Synchroniser 2 */ { 0x000001A3, 0x0000 }, /* R419 - FLL2 Synchroniser 3 */ { 0x000001A4, 0x0000 }, /* R420 - FLL2 Synchroniser 4 */ { 0x000001A5, 0x0000 }, /* R421 - FLL2 Synchroniser 5 */ { 0x000001A6, 0x0000 }, /* R422 - FLL2 Synchroniser 6 */ + { 0x000001A7, 0x0001 }, /* R422 - FLL2 Synchroniser 7 */ { 0x000001A9, 0x0000 }, /* R425 - FLL2 Spread Spectrum */ { 0x000001AA, 0x0004 }, /* R426 - FLL2 GPIO Clock */ { 0x00000200, 0x0006 }, /* R512 - Mic Charge Pump 1 */ @@ -1392,6 +1396,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL1_CONTROL_4: case ARIZONA_FLL1_CONTROL_5: case ARIZONA_FLL1_CONTROL_6: + case ARIZONA_FLL1_CONTROL_7: case ARIZONA_FLL1_LOOP_FILTER_TEST_1: case ARIZONA_FLL1_NCO_TEST_0: case ARIZONA_FLL1_SYNCHRONISER_1: @@ -1400,6 +1405,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL1_SYNCHRONISER_4: case ARIZONA_FLL1_SYNCHRONISER_5: case ARIZONA_FLL1_SYNCHRONISER_6: + case ARIZONA_FLL1_SYNCHRONISER_7: case ARIZONA_FLL1_SPREAD_SPECTRUM: case ARIZONA_FLL1_GPIO_CLOCK: case ARIZONA_FLL2_CONTROL_1: @@ -1408,6 +1414,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL2_CONTROL_4: case ARIZONA_FLL2_CONTROL_5: case ARIZONA_FLL2_CONTROL_6: + case ARIZONA_FLL2_CONTROL_7: case ARIZONA_FLL2_LOOP_FILTER_TEST_1: case ARIZONA_FLL2_NCO_TEST_0: case ARIZONA_FLL2_SYNCHRONISER_1: @@ -1416,6 +1423,7 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_FLL2_SYNCHRONISER_4: case ARIZONA_FLL2_SYNCHRONISER_5: case ARIZONA_FLL2_SYNCHRONISER_6: + case ARIZONA_FLL2_SYNCHRONISER_7: case ARIZONA_FLL2_SPREAD_SPECTRUM: case ARIZONA_FLL2_GPIO_CLOCK: case ARIZONA_MIC_CHARGE_PUMP_1: -- 1.7.2.5

2 4

[alsa-devel] [PATCH RESEND] mfd: arizona: Move regulator disable to after marking cache only
by Charles Keepax 07 Aug '13

07 Aug '13

If we disable DCVDD before we mark the cache as cache only, we might attempt to write to the chip whilst it is powered down and lose a write. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/arizona-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 89a1153..5ac3aa4 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -438,9 +438,9 @@ static int arizona_runtime_suspend(struct device *dev) } } - regulator_disable(arizona->dcvdd); regcache_cache_only(arizona->regmap, true); regcache_mark_dirty(arizona->regmap); + regulator_disable(arizona->dcvdd); return 0; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH v6 0/8] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 07 Aug '13

07 Aug '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) As the at91 chips are on their way to the common clock framework, I thought it may be better to retrieve the codec MCLK via this framework. So it needs Boris Brezillon patch set "ARM: at91: move to common clk framework" I've separated this patch because I don't know what is better: * applying the patchset with a hard coded clock, and once at91 has moved to common clock framework, update the code and change the DT binding. * wait for the common clock framework. Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. Maybe atmel_wm8731.c ? Since it could be merged with sam9g20_wm8731.c when the g20 is DT only (There will be some more work though). It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130729 + "ARM: at91: move to common clk framework" Changes from v5: * Update the wm8731 DT binding documentation with available audio endpoints. * Remove Uwe's signed-off since it seems to be "an integration signed-off" * Remove rate constrains if freq==0 in wm8731. * Allocate dynamically card, dai and private structures. * Add a separate patch to set the oscillator value via DT. NB: this last patch needs Boris Brezillon patch set "ARM: at91: move to common clk framework" Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Richard Genoud (7): ASoC: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration ASoC: sam9x5: get codec MCLK via device tree .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 39 ++++ Documentation/devicetree/bindings/sound/wm8731.txt | 9 + arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 32 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 218 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 60 +++++- 9 files changed, 392 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

3 20

[alsa-devel] [PATCH] ASoC: ad1980: Provide stub DAPM support
by Mark Brown 07 Aug '13

07 Aug '13

From: Mark Brown <broonie(a)linaro.org> Since non-DAPM devices are not going to be supported provide DAPM input and output widgets and hook them up to the DAIs. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ad1980.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 89fcf7d..7257a88 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -96,6 +96,44 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; +static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_INPUT("CD_L"), +SND_SOC_DAPM_INPUT("CD_R"), +SND_SOC_DAPM_INPUT("AUX_L"), +SND_SOC_DAPM_INPUT("AUX_R"), +SND_SOC_DAPM_INPUT("LINE_IN_L"), +SND_SOC_DAPM_INPUT("LINE_IN_R"), + +SND_SOC_DAPM_OUTPUT("LFE_OUT"), +SND_SOC_DAPM_OUTPUT("CENTER_OUT"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_L"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_R"), +SND_SOC_DAPM_OUTPUT("MONO_OUT"), +SND_SOC_DAPM_OUTPUT("HP_OUT_L"), +SND_SOC_DAPM_OUTPUT("HP_OUT_R"), +}; + +static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { + { "Capture", NULL, "MIC1" }, + { "Capture", NULL, "MIC2" }, + { "Capture", NULL, "CD_L" }, + { "Capture", NULL, "CD_R" }, + { "Capture", NULL, "AUX_L" }, + { "Capture", NULL, "AUX_R" }, + { "Capture", NULL, "LINE_IN_L" }, + { "Capture", NULL, "LINE_IN_R" }, + + { "LFE_OUT", NULL, "Playback" }, + { "CENTER_OUT", NULL, "Playback" }, + { "LINE_OUT_L", NULL, "Playback" }, + { "LINE_OUT_R", NULL, "Playback" }, + { "MONO_OUT", NULL, "Playback" }, + { "HP_OUT_L", NULL, "Playback" }, + { "HP_OUT_R", NULL, "Playback" }, +}; + static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -253,6 +291,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .reg_cache_step = 2, .write = ac97_write, .read = ac97_read, + + .dapm_widgets = ad1980_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets), + .dapm_routes = ad1980_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes), }; static int ad1980_probe(struct platform_device *pdev) -- 1.8.4.rc1

2 1

[alsa-devel] horribly wrong code when --with-versioned is active
by John Spencer 07 Aug '13

07 Aug '13

if --with-versioned is active (default), a couple of macros in pcm.c start generating some completely broken, __old-prefixed wrapper functions, which then are getting used whenever the actual function is called. for example: snd_pcm_hw_params_set_buffer_time_near __OLD_NEAR1(snd_pcm_hw_params_set_buffer_time_near, unsigned int); -> #define __OLD_NEAR1(name, ret_type) __P_OLD_NEAR1(__old_, name, ret_type) -> #define __P_OLD_NEAR1(pfx, name, ret_type) \ ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val, int *dir) \ { \ if (INTERNAL(name)(pcm, params, &val, dir) < 0) \ return 0; \ return (ret_type)val; \ } this will lead to generating a function __old_snd_pcm_hw_params_set_buffer_time_near which expands to unsigned int __old_snd_pcm_hw_params_set_buffer_time_near(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val, int *dir) { if snd1_pcm_hw_params_set_buffer_time_near(pcm, params, &val, dir) < 0) return 0; return (ret_type)val; } there 2 bugs in there, 1) the real function gets passed a pointer to a pointer of unsigned, which is then happily dereferenced and the original pointer used as an int, and 2) the return type logic is wrong, in case of a non-error, the original pointer will be returned instead of 0 to indicate success. the right fix would look something like this: @@ -7190,19 +7192,15 @@ __OLD_GET1(snd_pcm_hw_params_get_tick_time_max, unsigned int, unsigned int); #define __P_OLD_NEAR(pfx, name, ret_type) \ -ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val) \ +ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type *val) \ { \ - if (INTERNAL(name)(pcm, params, &val) < 0) \ - return 0; \ - return (ret_type)val; \ + return(INTERNAL(name)(pcm, params, val)); \ } #define __P_OLD_NEAR1(pfx, name, ret_type) \ -ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type val, int *dir) \ +ret_type pfx##name(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, ret_type *val, int *dir) \ { \ - if (INTERNAL(name)(pcm, params, &val, dir) < 0) \ - return 0; \ - return (ret_type)val; \ + return (INTERNAL(name)(pcm, params, val, dir) < 0); \ } #define __OLD_NEAR(name, ret_type) __P_OLD_NEAR(__old_, name, ret_type) this is only half of the fix though, the "old" getter functions seem to misbehave as well. the misbehaviour can be inspected by using a small openal-soft (version 1.14) example code, and breaking on alsa_reset_playback and single stepping through the invocation of CHECK(snd_pcm_hw_params_set_buffer_time_near(data->pcmHandle, hp, &bufferLen, NULL));

2 3

[alsa-devel] [PATCH] ASoC: imx-mc13783: Make SND_SOC_IMX_MC13783 visible again
by Fabio Estevam 07 Aug '13

07 Aug '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Commit 02502da45 (ASoC: imx-mc13783: Depend on ARCH_ARM) introduced 'ARCH_ARM' as a dependency for SND_SOC_IMX_MC13783, but this is a non-existent symbol. This makes the selection of SND_SOC_IMX_MC13783 to be impossible. Use the correct 'ARM' symbol instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index c26449b..e15f771 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -194,7 +194,7 @@ config SND_SOC_IMX_SGTL5000 config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" - depends on MFD_MC13783 && ARCH_ARM + depends on MFD_MC13783 && ARM select SND_SOC_IMX_SSI select SND_SOC_IMX_AUDMUX select SND_SOC_MC13783 -- 1.8.1.2

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume
by brian.austin＠cirrus.com 06 Aug '13

06 Aug '13

From: Brian Austin <brian.austin(a)cirrus.com> Beep Volume Min/Max was backwards. Change to SOC_SONGLE_SX_TLV for correct volume representation Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728..ee25f32 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

2 2

[alsa-devel] [PATCH 0/5] ASoC: R-Car sound DMA transfer support
by Kuninori Morimoto 06 Aug '13

06 Aug '13

Hi Mark These are DMAEngine support patches for R-Car sound Last patch cleanups empty SCU functions. I will work for regmap support soon, but I need more time for it. This patch doesn't use soc-dmaengine-pcm.c method, becouse current R-Car chip doesn't suuport dmaengine_prep_dma_cyclic(). Of course I can re-work to use it if chip/rcar-dmaengine supports it. these are based on mark/topic/rcar branch Kuninori Morimoto (5): ASoC: rsnd: remove platform dai and add dai_id on platform setting ASoC: rsnd: add common DMAEngine method ASoC: rsnd: SSI supports DMA transfer ASoC: rsnd: SSI supports DMA transfer via BUSIF ASoC: rsnd: scu: cleanup empty functions include/sound/rcar_snd.h | 25 ++++-- sound/soc/sh/rcar/core.c | 192 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/gen.c | 20 +++-- sound/soc/sh/rcar/rsnd.h | 44 +++++++++++ sound/soc/sh/rcar/scu.c | 163 ++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 150 ++++++++++++++++++++++++++++++++++-- 6 files changed, 529 insertions(+), 65 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 1/3] ASoC: tlv320aic26: Fix keyclick feature
by Lars-Peter Clausen 06 Aug '13

06 Aug '13

The tlv320aic26 contains a embedded snd_soc_codec struct which is referenced in the keyclick code. That struct is never initialized though, replace the embedded struct with a pointer and use that in the keyclick code. Cc: Grant Likely <grant.likely(a)secretlab.ca> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/tlv320aic26.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b1f6982..b192cd4 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,7 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; - struct snd_soc_codec codec; + struct snd_soc_codec *codec; int master; int datfm; int mclk; @@ -330,7 +330,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -346,9 +346,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -360,8 +360,11 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); */ static int aic26_probe(struct snd_soc_codec *codec) { + struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, err, i, reg; + aic26->codec = codec; + dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ -- 1.8.0

3 6

[alsa-devel] [PATCH] ASoC: wm5110: Correct input OSR bits for wm5110
by Charles Keepax 06 Aug '13

06 Aug '13

The input OSR bits are specified differently for wm5110 than for current revs of wm5102. This patch corrects support for this on wm5110. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 20 ++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 12 ++++-------- 3 files changed, 25 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8dc6881..779a0ee 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -553,6 +553,26 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static const char * const arizona_in_dmic_osr_text[] = { + "1.536MHz", "3.072MHz", "6.144MHz", +}; + +const struct soc_enum arizona_in_dmic_osr[] = { + SOC_ENUM_SINGLE(ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN2L_CONTROL, ARIZONA_IN2_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), +}; +EXPORT_SYMBOL_GPL(arizona_in_dmic_osr); + static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index fe1b794..b6b6d70 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -198,6 +198,7 @@ extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; extern const struct soc_enum arizona_ng_hold; +extern const struct soc_enum arizona_in_dmic_osr[]; extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fc41037..77fd531 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -58,14 +58,10 @@ static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0) static const struct snd_kcontrol_new wm5110_snd_controls[] = { -SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, - ARIZONA_IN4_OSR_SHIFT, 1, 0), +SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), +SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), +SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Fix class W controls
by Mark Brown 06 Aug '13

06 Aug '13

From: Mark Brown <broonie(a)linaro.org> Commit 6e0650 (ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it) went too far and converted a DAPM control to use SOC_SINGLE_EXT() which crashes. Revert that portion of the patch. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d76b056..29ad480 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1432,8 +1432,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, }; #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ - SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ - snd_soc_get_volsw, wm8994_put_class_w) + SOC_SINGLE_EXT(xname, reg, shift, 1, 0, \ + snd_soc_dapm_get_volsw, wm8994_put_class_w) static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- 1.8.4.rc0

2 2

[alsa-devel] [PATCH v2 0/9] ALSA: usb-audio: Refactor set_format()
by Eldad Zack 06 Aug '13

06 Aug '13

Hi Takashi, This is a low-priority series (applies against current for-next). This series refactors set_format() and moves the sync endpoint initilization code into separate function. The quirks for implicit feedback are moved to a separate function as well. The intended result is code that is easier to read without function changes. Changes from v2: * Cleaned up the patchset, indentations and whitespace fixes. * Patch #2: removed early returns when subs->sync_endpoint is already set. This change now follows original semantics. * Patch #9: I first corrected it thanks to Clemens Ladisch's feedback, but then I wasn't sure if it improves anything, so I dropped it entirely. * Dropped patch #10. Instead, added patch (now #9) to WARN_ON and return NULL in snd_usb_add_endpoint. Otherwise, the check should be done at each call site, and it "shouldn't happen" anyway. Cheers, Eldad v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-July/064268.html Eldad Zack (9): ALSA: usb-audio: remove disabled debug code in set_format ALSA: usb-audio: remove assignment from if condition ALSA: usb-audio: separate sync endpoint setting from set_format ALSA: usb-audio: move implicit fb quirks to separate function ALSA: usb-audio: reverse condition logic in set_sync_endpoint ALSA: usb-audio: do not initialize and check implicit_fb ALSA: usb-audio: remove is_playback from implicit feedback quirks ALSA: usb-audio: remove implicit_fb from quirk ALSA: usb-audio: WARN_ON when alts is passed as NULL sound/usb/endpoint.c | 3 + sound/usb/pcm.c | 243 +++++++++++++++++++++++++++++---------------------- 2 files changed, 141 insertions(+), 105 deletions(-) -- 1.8.1.5

2 8

[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume
by Brian Austin 06 Aug '13

06 Aug '13

Beep Volume Min/Max was backwards. Change to SOC_SONGLE_SX_TLV for correct volume representation Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728..ee25f32 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

4 4

[alsa-devel] [PATCH] ASoC: wm8994: Fix class W controls
by Mark Brown 05 Aug '13

05 Aug '13

From: Mark Brown <broonie(a)linaro.org> Commit 6e0650 (ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it) went too far and converted a DAPM control to use SOC_SINGLE_EXT() which crashes. Revert that portion of the patch. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8994.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d76b056..f32998a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1432,8 +1432,10 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, }; #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ - SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ - snd_soc_get_volsw, wm8994_put_class_w) +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = wm8994_put_class_w, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- 1.8.4.rc0

2 2

[alsa-devel] [PATCH 2/2] ASoC: cs42l52: Add new TLV for Beep Volume
by Brian Austin 05 Aug '13

05 Aug '13

CS42L52 Beep control uses 2dB scale from -56dB Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index ee25f32..1e7d03f 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, + 0, 0x07, 0x1f, beep_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: cs42l52: Fixes for Beep Volume kcontrol (fwd)
by Brian Austin 05 Aug '13

05 Aug '13

This patch reorders the min/max values for BEEP Volume. Also changes SOC_SINGLE_TLV to SOC_SINGLE_SX_TLV Add DB_SCALE beep_tlv for volume control Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0f6f481..66d1e42 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(beep_tlv, -5600, 200, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -449,7 +451,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", + CS42L52_BEEP_VOL, 0, 0x07, 0x1f, beep_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- 1.7.10.4

2 6

[alsa-devel] [PATCH 1/3] ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture
by Lothar Waßmann 05 Aug '13

05 Aug '13

When a sound capture/playback is terminated while a playback/capture is running, power_vag_event() will clear SGTL5000_CHIP_ANA_POWER in the SND_SOC_DAPM_PRE_PMD event, thus muting the respective other channel. Don't clear SGTL5000_CHIP_ANA_POWER when both DAC and ADC are active to prevent this. Signed-off-by: Lothar Waßmann <LW(a)KARO-electronics.de> --- sound/soc/codecs/sgtl5000.c | 16 +++++++++++++--- 1 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8b9090..a4b2973 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; + switch (event) { case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, @@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); + /* + * Don't clear VAG_POWERUP, when both DAC and ADC are + * operational to prevent inadvertently starving the + * other one of them. + */ + if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) & + mask) != mask) { + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + } break; default: break; -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: pcm1792a: Fix build with !OF
by Mark Brown 05 Aug '13

05 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1792a.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3f83bf9..72cf835 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -169,13 +169,11 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .ops = &pcm1792a_dai_ops, }; -#ifdef CONFIG_OF static const struct of_device_id pcm1792a_of_match[] = { { .compatible = "ti,pcm1792a", }, { } }; MODULE_DEVICE_TABLE(of, pcm1792a_of_match); -#endif static const struct regmap_config pcm1792a_regmap = { .reg_bits = 8, @@ -231,7 +229,7 @@ static struct spi_driver pcm1792a_codec_driver = { .driver = { .name = "pcm1792a", .owner = THIS_MODULE, - .of_match_table = pcm1792a_of_match, + .of_match_table = of_match_ptr(pcm1792a_of_match), }, .id_table = pcm1792a_spi_ids, .probe = pcm1792a_spi_probe, -- 1.8.4.rc0

1 0

[alsa-devel] [RFC PATCH] ASoC: Add PCM1792A spi mode codec support
by Michael Trimarchi 05 Aug '13

05 Aug '13

Add PCM1792A spi mode codec support. This version implements only a subset of functionalities. Tested connect to a pandaboard ES device and based on recently pcm1681 codec. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- .../devicetree/bindings/sound/pcm1792a.txt | 18 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1792a.c | 245 ++++++++++++++++++++ sound/soc/codecs/pcm1792a.h | 26 +++ 5 files changed, 295 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h diff --git a/Documentation/devicetree/bindings/sound/pcm1792a.txt b/Documentation/devicetree/bindings/sound/pcm1792a.txt new file mode 100644 index 0000000..b8e48e8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm1792a.txt @@ -0,0 +1,18 @@ +Texas Instruments pcm1792a DT bindings + +This driver supports the SPI bus. + +Required properties: + + - compatible: "ti,pcm1792a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Examples: + + codec_spi: 1792a@0 { + compatible = "ti,pcm1792a"; + spi-max-frequency = <600000>; + }; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fb..45a9e3c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 + select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -295,6 +296,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_PCM3008 tristate +config SND_SOC_PCM1792A + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd806..1f8ee0b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -43,6 +43,7 @@ snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -172,6 +173,7 @@ obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c new file mode 100644 index 0000000..99a4ee2 --- /dev/null +++ b/sound/soc/codecs/pcm1792a.c @@ -0,0 +1,245 @@ +/* + * PCM1792A ASoC codec driver + * + * Copyright (c) Amarula Solutions B.V. 2013 + * + * Michael Trimarchi <michael(a)amarulasolutions.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; either version 2 + * of the License, or (at your option) any later version. + * + * 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/slab.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/spi/spi.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/of_device.h> + +#include "pcm1792a.h" + +#define PCM1792A_DAC_VOL_LEFT 0x10 +#define PCM1792A_DAC_VOL_RIGHT 0x11 +#define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL + +#define PCM1792A_FMT_MASK 0x70 +#define PCM1792A_FMT_SHIFT 4 +#define PCM1792A_MUTE_MASK 0x01 +#define PCM1792A_MUTE_SHIFT 0 +#define PCM1792A_ATLD_ENABLE (1 << 7) + +static const struct reg_default pcm1792a_reg_defaults[] = { + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x50 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x01 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg) +{ + return reg >= 0x10 && reg <= 0x17; +} + +static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg) +{ + bool accessible; + + accessible = pcm1792a_accessible_reg(dev, reg); + + return accessible && reg != 0x16 && reg != 0x17; +} + +struct pcm1792a_private { + struct regmap *regmap; + unsigned int format; + unsigned int rate; +}; + +static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->format = format; + + return 0; +} + +static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + PCM1792A_MUTE_MASK, !!mute); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1792a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x02; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x05; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x04; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; + + ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_soc_dai_ops pcm1792a_dai_ops = { + .set_fmt = pcm1792a_set_dai_fmt, + .hw_params = pcm1792a_hw_params, + .digital_mute = pcm1792a_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1); + +static const struct snd_kcontrol_new pcm1792a_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, + PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, + pcm1792a_dac_tlv), +}; + +static struct snd_soc_dai_driver pcm1792a_dai = { + .name = "pcm1792a-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM1792A_RATES, + .formats = PCM1792A_FORMATS, }, + .capture = { + .channels_min = 0, + .channels_max = 0, + }, + .ops = &pcm1792a_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1792a_of_match[] = { + { .compatible = "ti,pcm1792a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1792a_of_match); +#endif + +static const struct regmap_config pcm1792a_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 24, + .reg_defaults = pcm1792a_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), + .writeable_reg = pcm1792a_writeable_reg, + .readable_reg = pcm1792a_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { + .controls = pcm1792a_controls, + .num_controls = ARRAY_SIZE(pcm1792a_controls), +}; + +static int pcm1792a_spi_probe(struct spi_device *spi) +{ + struct pcm1792a_private *pcm1792a; + int ret; + + pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private), + GFP_KERNEL); + if (!pcm1792a) + return -ENOMEM; + + spi_set_drvdata(spi, pcm1792a); + + pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap); + if (IS_ERR(pcm1792a->regmap)) { + ret = PTR_ERR(pcm1792a->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&spi->dev, + &soc_codec_dev_pcm1792a, &pcm1792a_dai, 1); +} + +static int pcm1792a_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm1792a_spi_ids[] = { + { "pcm1792a", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids); + +static struct spi_driver pcm1792a_codec_driver = { + .driver = { + .name = "pcm1792a", + .owner = THIS_MODULE, + .of_match_table = pcm1792a_of_match, + }, + .id_table = pcm1792a_spi_ids, + .probe = pcm1792a_spi_probe, + .remove = pcm1792a_spi_remove, +}; + +module_spi_driver(pcm1792a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM1792A driver"); +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h new file mode 100644 index 0000000..7a83d1f --- /dev/null +++ b/sound/soc/codecs/pcm1792a.h @@ -0,0 +1,26 @@ +/* + * definitions for PCM1792A + * + * Copyright 2013 Amarula Solutions + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * 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. + */ + +#ifndef __PCM1792A_H__ +#define __PCM1792A_H__ + +#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE) + +#endif -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: core: init delayed_work for codec-codec links
by Richard Fitzgerald 05 Aug '13

05 Aug '13

If soc_probe_link_dais() finds a codec-codec link it skips creating a compress or pcm stream and links the DAIs together. But it must also init the delayed_work otherwise shutting down the DAI chain will fault when calling flush_delayed_work_sync() on the linked DAI. Pointing it to a dummy work callback is cleaner than taking special cases in the code to bypass the flush_delayed_work_sync(). Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4489c5b..bbe136c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -199,6 +199,10 @@ static ssize_t pmdown_time_set(struct device *dev, return count; } +static void dummy_delayed_work(struct work_struct *work) +{ +} + static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); #ifdef CONFIG_DEBUG_FS @@ -1428,6 +1432,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return ret; } } else { + INIT_DELAYED_WORK(&rtd->delayed_work, dummy_delayed_work); + /* link the DAI widgets */ play_w = codec_dai->playback_widget; capture_w = cpu_dai->capture_widget; -- 1.7.2.5

2 11

[alsa-devel] [PATCH] sound/sgtl5000: fix codec clock source transition to avoid clockless moment
by Oskar Schirmer 05 Aug '13

05 Aug '13

Powering down PLL before switching to a mode that does not use it is a bad idea. It would cause the SGTL5000 be without internal clock supply, especially on the I2C interface, which would make subsequent access to it fail. Thus, in case of not using PLL any longer, first set the mode control, then power down PLL. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/codecs/sgtl5000.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6c8a9e7..8ecb201 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -644,16 +644,19 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); } else { + /* otherwise, clk_ctrl must be set before pll power down */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + /* power down pll */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, 0); } - /* if using pll, clk_ctrl must be set after pll power up */ - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); - return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: dapm: Implement mixer input auto-disable
by Lars-Peter Clausen 05 Aug '13

05 Aug '13

Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Rebased onto the latest asoc/topic/dapm --- include/sound/soc-dapm.h | 17 +++++++- include/sound/soc.h | 28 ++++++------ sound/soc/soc-dapm.c | 108 +++++++++++++++++++++++++++++++++++++++++------ 3 files changed, 125 insertions(+), 28 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3575721..c728d28 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -280,14 +280,26 @@ struct device; { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_AUTODISABLE(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_AUTODISABLE(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ @@ -484,6 +496,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_in, /* link to DAI structure */ snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ + snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ }; enum snd_soc_dapm_subclass { diff --git a/include/sound/soc.h b/include/sound/soc.h index b1e1f96..6201c6e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -30,13 +30,13 @@ /* * Convenience kcontrol builders */ -#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert) \ +#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ .rshift = shift_right, .max = xmax, .platform_max = xmax, \ - .invert = xinvert}) -#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ - SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert) + .invert = xinvert, .autodisable = xautodisable}) +#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ + SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert}) @@ -52,7 +52,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ @@ -68,7 +68,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -97,7 +97,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw, \ @@ -119,7 +119,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -190,14 +190,14 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = \ - SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) } + SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -206,7 +206,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -216,7 +216,7 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \ - xmax, xinvert) } + xmax, xinvert, 0) } #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -1088,7 +1088,9 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift, invert; + unsigned int reg, rreg, shift, rshift; + unsigned int invert:1; + unsigned int autodisable:1; }; struct soc_bytes { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5f64c16..0944bc4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -47,6 +47,15 @@ #define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; +static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, + const char *control, + int (*connected)(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink)); +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -73,16 +82,18 @@ static int dapm_up_seq[] = { [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, [snd_soc_dapm_line] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_kcontrol] = 11, + [snd_soc_dapm_post] = 12, }; static int dapm_down_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_adc] = 1, - [snd_soc_dapm_hp] = 2, - [snd_soc_dapm_spk] = 2, - [snd_soc_dapm_line] = 2, - [snd_soc_dapm_out_drv] = 2, + [snd_soc_dapm_kcontrol] = 1, + [snd_soc_dapm_adc] = 2, + [snd_soc_dapm_hp] = 3, + [snd_soc_dapm_spk] = 3, + [snd_soc_dapm_line] = 3, + [snd_soc_dapm_out_drv] = 3, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, @@ -176,6 +187,7 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( struct dapm_kcontrol_data { unsigned int value; + struct snd_soc_dapm_widget *widget; struct list_head paths; struct snd_soc_dapm_widget_list *wlist; }; @@ -184,6 +196,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data; + struct soc_mixer_control *mc; data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) { @@ -195,6 +208,39 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, INIT_LIST_HEAD(&data->paths); + switch (widget->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + mc = (struct soc_mixer_control *)kcontrol->private_value; + + if (mc->autodisable) { + struct snd_soc_dapm_widget template; + + memset(&template, 0, sizeof(template)); + template.reg = mc->reg; + template.mask = (1 << fls(mc->max)) - 1; + template.shift = mc->shift; + if (mc->invert) + template.off_val = mc->max; + else + template.off_val = 0; + template.on_val = template.off_val; + template.id = snd_soc_dapm_kcontrol; + template.name = kcontrol->id.name; + + data->widget = snd_soc_dapm_new_control(widget->dapm, + &template); + if (!data->widget) { + kfree(data); + return -ENOMEM; + } + } + break; + default: + break; + } + kcontrol->private_data = data; return 0; @@ -203,6 +249,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + kfree(data->widget); kfree(data->wlist); kfree(data); } @@ -246,6 +293,21 @@ static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol, struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); list_add_tail(&path->list_kcontrol, &data->paths); + + if (data->widget) { + snd_soc_dapm_add_path(data->widget->dapm, data->widget, + path->source, NULL, NULL); + } +} + +static bool dapm_kcontrol_is_powered(const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + if (!data->widget) + return true; + + return data->widget->power; } static struct list_head *dapm_kcontrol_get_path_list( @@ -275,6 +337,9 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; + if (data->widget) + data->widget->on_val = value; + data->value = value; return true; @@ -515,6 +580,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_spk: case snd_soc_dapm_line: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -880,6 +946,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -975,6 +1042,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -1523,7 +1591,7 @@ static void dapm_widget_update(struct snd_soc_card *card) unsigned int wi; int ret; - if (!update) + if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; wlist = dapm_kcontrol_get_wlist(update->kcontrol); @@ -1668,6 +1736,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -2335,6 +2404,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2717,6 +2787,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2724,17 +2795,24 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + unsigned int val; if (snd_soc_volsw_is_stereo(mc)) dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); - ucontrol->value.integer.value[0] = - (snd_soc_read(codec, reg) >> shift) & mask; + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + if (dapm_kcontrol_is_powered(kcontrol)) + val = (snd_soc_read(codec, reg) >> shift) & mask; + else + val = dapm_kcontrol_get_value(kcontrol); + mutex_unlock(&card->dapm_mutex); + if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = max - val; + else + ucontrol->value.integer.value[0] = val; return 0; } @@ -2775,11 +2853,14 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - mask = mask << shift; - val = val << shift; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_kcontrol_set_value(kcontrol, val); + + mask = mask << shift; + val = val << shift; + change = snd_soc_test_bits(codec, reg, mask, val); if (change) { update.kcontrol = kcontrol; @@ -3179,6 +3260,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: w->power_check = dapm_supply_check_power; break; default: -- 1.8.0

2 1

[alsa-devel] [PATCH v3 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 05 Aug '13

05 Aug '13

Changelog: v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 62 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1311 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 227 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1781 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

2 4

[alsa-devel] ASoC: omap-mcbsp 192000 freq
by Michael Trimarchi 05 Aug '13

05 Aug '13

Hi I'm working on having pcm1792a and omap platform. Right now I have already sent and rfc for the codec and I have a local build that change this file too to support 192000. What I don't understand is the meaning of: .period_bytes_min = 32, What is the relation between fifo_size and period_bytes_min? because at this frequency 32 bytes are not so much and the fifo of the port are >= 512 bytes. Is it possible to increase the period_bytes_min without violating the hw constraints? Michael .period_bytes_max = 64 * 1024, .periods_min = 2, .periods_max = 255, .buffer_bytes_max = 128 * 1024, >From f1e6fcfd81e6642ba5695c848ac6aa729e7467aa Mon Sep 17 00:00:00 2001 From: Michael Trimarchi <michael(a)amarulasolutions.com> Date: Mon, 5 Aug 2013 12:26:35 +0200 Subject: [PATCH] OMAP MCBSP can support 192000 frame rate. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..d285ddd 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,7 +39,7 @@ #include "mcbsp.h" #include "omap-mcbsp.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000) #define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ xhandler_get, xhandler_put) \ -- 1.7.9.5

1 0

[alsa-devel] (no subject)
by Chih-Chung Chang 05 Aug '13

05 Aug '13

This is a revised patch which also checks if the mic is external.

2 2

[alsa-devel] [PATCH 1/1] ALSA: hda - Fix jack gating when auto_{mute, mic} is suppressed.
by Chih-Chung Chang 05 Aug '13

05 Aug '13

The snd_hda_jack_set_gating_jack() call didn't work when auto_{mute,mic} is suppressed because (1) am_entry is not filled with nid of the mic pin. (2) The jacks are not created (by snd_hda_jack_detect_enable_callback) before the snd_hda_jack_set_gating_jack call. Now we use the first input pin nid directly, and create the jack if it doesn't exist yet. Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973..aa02c0f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -247,8 +247,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid) { - struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid); - struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid); + struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid); + struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid); if (!gated || !gating) return -EINVAL; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..2b64de5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3258,11 +3258,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PROBE) { - if (snd_BUG_ON(!spec->gen.am_entry[1].pin || - !spec->gen.autocfg.hp_pins[0])) + int mic_pin = spec->gen.autocfg.inputs[0].pin; + int hp_pin = spec->gen.autocfg.hp_pins[0]; + + if (snd_BUG_ON(!mic_pin || !hp_pin)) return; - snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin, - spec->gen.autocfg.hp_pins[0]); + snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin); } } -- 1.8.3

3 2

[alsa-devel] [PATCH v2 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 05 Aug '13

05 Aug '13

Changelog: v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 62 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1311 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 227 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1781 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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Re: [alsa-devel] hda: sink info in case of AMD Radeon GPU card (no ELD?)
by Pierre Ossman 05 Aug '13

05 Aug '13

On Wed, 17 Apr 2013 14:04:00 +0200 Rafał Miłecki <zajec5(a)gmail.com> wrote: > > I tried hacking this sink info problem again. I've installed "fglrx" > on my AMD Radeon HD 6970M (BARTS == DCE5) and verified it sets SAD > registers: Any progress on this? I fiddled around with providing the ELD stuff out-of-band, but that's apparently not sufficient. I guess the widget capabilities property also needs to change. Did you check to see if that happened with fglrx? Rgds -- -- Pierre Ossman WARNING: This correspondence is being monitored by FRA, a Swedish intelligence agency. Make sure your server uses encryption for SMTP traffic and consider using PGP for end-to-end encryption.

2 1

[alsa-devel] [PATCH] ASoC: rt5640: change widget sequence for depop
by bardliao＠realtek.com 05 Aug '13

05 Aug '13

From: Bard Liao <bardliao(a)realtek.com> Add mute/unmute control in widget event and modify the power on/off sequence to avoid pop noise. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 387 ++++++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/rt5640.h | 15 ++ 2 files changed, 352 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..3490f38 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -378,21 +378,119 @@ static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; +static const char * const rt5640_mute_mode[] = {"mute", "unmute"}; + +static const SOC_ENUM_SINGLE_DECL(rt5640_hp_enum, 0, 0, rt5640_mute_mode); +static const SOC_ENUM_SINGLE_DECL(rt5640_spo_enum, 0, 0, rt5640_mute_mode); + static const SOC_ENUM_SINGLE_DECL( rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +static int rt5640_spo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->spo_l_mute; + ucontrol->value.integer.value[1] = !rt5640->spo_r_mute; + + return 0; +} + +static int rt5640_spo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_DIG1) & RT5640_PWR_CLS_D; + + if (ucontrol->value.integer.value[0]) { + rt5640->spo_l_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->spo_l_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, + RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->spo_r_mute = false; + if (pow) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->spo_r_mute = true; + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + } + + return 0; +} + +static int rt5640_hp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = !rt5640->hp_l_mute; + ucontrol->value.integer.value[1] = !rt5640->hp_r_mute; + + return 0; +} + +static int rt5640_hp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int pow; + + pow = snd_soc_read(codec, RT5640_PWR_ANLG1); + + if (ucontrol->value.integer.value[0]) { + rt5640->hp_l_mute = false; + if (pow & RT5640_PWR_HP_L) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, 0); + } else { + rt5640->hp_l_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + } + if (ucontrol->value.integer.value[1]) { + rt5640->hp_r_mute = false; + if (pow & RT5640_PWR_HP_R) + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, 0); + } else { + rt5640->hp_r_mute = true; + snd_soc_update_bits(codec, RT5640_HP_VOL, + RT5640_R_MUTE, RT5640_R_MUTE); + } + + + return 0; +} + static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("Speaker Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_spo_get, rt5640_spo_put), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_EXT("HP Playback Switch", SND_SOC_NOPM, + 0, 1, 1, 0, rt5640_hp_get, rt5640_hp_put), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -868,33 +966,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +1014,216 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_spo_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_r_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_R_MUTE, + RT5640_R_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_spo_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->spo_l_mute) + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_SPK_VOL, + RT5640_L_MUTE, RT5640_L_MUTE); + break; + + default: + return 0; + } + + return 0; +} + +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, + RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | + RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(20000, 25000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, + RT5640_PWR_FV1 | RT5640_PWR_FV2); + + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CO_MASK | + RT5640_HP_SG_MASK, + RT5640_HP_CO_EN | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, pow; + + regmap_read(rt5640->regmap, RT5640_PWR_ANLG1, &pow); + if (rt5640->hp_l_mute || !(pow & RT5640_PWR_HP_L)) + val |= RT5640_L_MUTE; + if (rt5640->hp_r_mute || !(pow & RT5640_PWR_HP_R)) + val |= RT5640_R_MUTE; + /* headphone unmute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK, RT5640_SMT_TRIG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK, RT5640_RSTN_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTN_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_RSTN_DIS | RT5640_HP_L_SMT_EN | + RT5640_HP_R_SMT_EN); + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, val); + usleep_range(40000, 45000); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | RT5640_HP_R_SMT_DIS); + +} + +static void rt5640_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* headphone mute sequence */ + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | + RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_96_KHZ << RT5640_CP_FQ3_SFT)); + regmap_write(rt5640->regmap, + RT5640_PR_BASE + RT5640_MAMP_INT_REG2, 0xfc00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK, RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK, RT5640_RSTP_EN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_RSTP_MASK | RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, RT5640_RSTP_DIS | + RT5640_HP_L_SMT_EN | RT5640_HP_R_SMT_EN); + + regmap_update_bits(rt5640->regmap, RT5640_HP_VOL, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + usleep_range(40000, 45000); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_SG_MASK | + RT5640_HP_L_SMT_MASK | + RT5640_HP_R_SMT_MASK, + RT5640_HP_SG_DIS | + RT5640_HP_L_SMT_DIS | + RT5640_HP_R_SMT_DIS); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_SMT_TRIG_MASK | + RT5640_HP_CD_PD_MASK | + RT5640_HP_CO_MASK | + RT5640_HP_CP_MASK | + RT5640_HP_SG_MASK | + RT5640_HP_CB_MASK, + RT5640_SMT_TRIG_DIS | + RT5640_HP_CD_PD_EN | + RT5640_HP_CO_DIS | RT5640_HP_CP_PD | + RT5640_HP_SG_EN | RT5640_HP_CB_PD); +} + +static int rt5640_hp_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: + rt5640_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_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: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1413,23 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("SPO L Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_l_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("SPO R Amp", SND_SOC_NOPM, + 0, 0, NULL, 0, rt5640_spo_r_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1396,13 +1685,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"SPO L Amp", NULL, "SPOL MIX"}, + {"SPO R Amp", NULL, "SPOR MIX"}, + {"SPOLP", NULL, "SPO L Amp"}, + {"SPOLN", NULL, "SPO L Amp"}, + {"SPORP", NULL, "SPO R Amp"}, + {"SPORN", NULL, "SPO R Amp"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,7 +1703,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, + {"HPOL", NULL, "HP Amp"}, {"HPOL", NULL, "HP L Amp"}, + {"HPOR", NULL, "HP Amp"}, {"HPOR", NULL, "HP R Amp"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, @@ -1792,17 +2085,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(15000, 20000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +2135,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..cb6bce9 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,10 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool spo_l_mute; + bool spo_r_mute; + bool hp_l_mute; + bool hp_r_mute; }; #endif -- 1.8.1.1.439.g50a6b54

3 2

Re: [alsa-devel] [PATCH v3 4/4] ASoC: kirkwood: change kirkwood-i2s to mvebu-pcm-audio and DT doc
by Russell King - ARM Linux 04 Aug '13

04 Aug '13

On Wed, Jul 31, 2013 at 08:18:58AM +0200, Jean-Francois Moine wrote: > diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c > index 9844010..4f817a2 100644 > --- a/sound/soc/kirkwood/kirkwood-i2s.c > +++ b/sound/soc/kirkwood/kirkwood-i2s.c > @@ -25,7 +25,7 @@ > > #include "kirkwood.h" > > -#define DRV_NAME "kirkwood-i2s" > +#define DRV_NAME "mvebu-pcm-audio" This breaks non-DT users of this driver. The device in arch/arm/mach-kirkwood is left as "kirkwood-i2s" but the platform device is now called something different. > static struct of_device_id kirkwood_i2s_of_match[] = { > - { .compatible = "marvell,kirkwood-i2s" }, > + { .compatible = "marvell,mvebu-pcm-audio" }, You've only just introduced "marvell,kirkwood-i2s" in the previous patch. Why introduce something and then immediately change its name? This makes no sense what so ever.

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Re: [alsa-devel] [PATCH v3 2/4] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Russell King - ARM Linux 04 Aug '13

04 Aug '13

On Wed, Jul 31, 2013 at 08:18:06AM +0200, Jean-Francois Moine wrote: > To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, > this patch merges the kirkwood-i2c and kirkwood-dma drivers into one > module associated with 'kirkwood-i2s'. I suggest holding off on this stuff at the moment. I think Mark and Liam (who now have a whole raft of emails from me today) have some work to do to fix ASoC to work how they're saying it should work, because to me ASoC seems to contradict what they're telling me. To put it another way, it must be buggy. The DAPM stuff seems to be the worst thing since mouldy bread. I'm chasing what seems to be multiple bugs through this stuff, and many of them are not particularly nice. For example - we register multiple copies of DAPM widgets for the same set of declarations if both a CPU DAI and a platform share the same struct device, and thereby end up overwriting some pointers in the DAI structure. It seems that CPU DAIs themselves aren't supposed to have DAPM widgets, but the core creates some... but there's no explicit cleanup of them unlike the other DAPMs, and there's no debugfs for them. The bias levels are stuck at off/standby all the time, which makes any kind of PM with spdif-dit impossible - or even any routing between stream input and output. Basically, I'm tearing my hair out today looking at this stuff and getting nowhere fast - all I'm doing is finding more and more problems. For example, where a codec has no input/output widgets declared, it used to be powered up automatically by ASoC as a matter of course. Things like UDA134x and other things. Things like spdif-dit. That "mysteriously" stopped happening. ASoC: dapm: Treat DAI widgets like AIF widgets for power seems to be the cause, this results in such setups having zero connected inputs/outputs reported, which causes them to remain powered down - because what used to happen before was the DAI links would report their powered state depending on whether they were active, and they are set active in soc_dapm_stream_event(). Now, when playback widget for the Codec and CPU DAIs get marked as active. The playback widget is created as a snd_soc_dapm_dai_in. It's power check function is set to dapm_dac_check_power. Since this widget is active, it checks for connected outputs via is_connected_output_ep(). We drop through to the second switch statement (why do we have two there? They're both switching on the same damned variable and its not like a widget can change its ID.) This is where the different behaviour has appeared - when these were just a simple snd_soc_dapm_dai, we used to just do the snd_soc_dapm_suspend_check() here, but we don't anymore. This is a snd_soc_dapm_dai_in type of widget, so we fall through this switch statement now and start searching the paths. As these codecs have no paths, this ultimately ends up returning no connections. Hence why these codecs with no DAPM widgets declared but have PM support via the bias level stuff have stopped working. Now, about the spdif-dit, if we're going to have to add "pin" widgets to it, what the output of a SPDIF in terms of DAPM widgets? At a guess, it's a "codec pin" despite there being no codec and no "pin" on that codec in reality, and that "pin" is always active. With that "fixed" (rather, altered to a state where it behaves the same way as it used to before the above commit) if I set up a DAPM route between the CPU DAI playback stream, an AIF (for spdif output) and the spdif-dit playback stream, I see the playback streams marked active and powered up, but the AIF stream remains powered down: spdif-dit/dapm/Playback:Playback: On in 1 out 1 spdif-dit/dapm/Playback: stream Playback active spdif-dit/dapm/Playback: in "static" "spdif-playback" spdif-dit/dapm/bias_level:On kirkwood-audio.1/dapm/spdif-playback:spdif-playback: Off in 1 out 0 kirkwood-audio.1/dapm/spdif-playback: in "static" "dma-playback" kirkwood-audio.1/dapm/spdif-playback: out "static" "Playback" kirkwood-audio.1/dapm/dma-playback:dma-playback: On in 1 out 1 kirkwood-audio.1/dapm/dma-playback: stream dma-playback active kirkwood-audio.1/dapm/dma-playback: out "static" "spdif-playback" kirkwood-audio.1/dapm/dma-playback: out "static" "i2s-playback" It looks to me like the DAPM stuff is - in one plain and simple word - buggered. I've no idea what the right fixes are in this area. It needs someone like Mark or Liam who supposedly understand this to spend time checking that it actually operates as they _think_ it should operate, because at the moment it plainly doesn't.

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[alsa-devel] [PATCH V2 0/8] Add audio support on smdk5420
by Padmavathi Venna 04 Aug '13

04 Aug '13

Samsung has different versions of I2S introduced in different platforms. Each version has some new support added for multichannel, secondary fifo, s/w reset control, internal mux for rclk src clk and tdm support. Each newly added change has a quirk. So this patch adds all the required quirks as driver data and based on compatible string from dtsi fetches the quirks. This also adds i2s support on exynos5420 and make relevent changes in the dtsi files. Changes since V1: - Pass quirks as driver data and fetch the quirks based on compatible string from dtsi file as suggested by Tomasz Figa and Mark Brown - Make the I2S driver more flexible with respect to register access as suggested by Tomasz Figa and Mark Brown - Add 5420 support in the driver. - Modify the dtsi files with the corresponding compatible strings and removed the i2s quirks from 5250 dtsi file. - Updated the i2s Documentation with relevent changes and i2s versioning info. - Add i2s nodes on exynos5420.dtsi - Enable sound support on smdk5420 This patch set is dependent on the following dma and audio subsystem clk controller patches. http://comments.gmane.org/gmane.linux.kernel.samsung-soc/20661 http://comments.gmane.org/gmane.linux.kernel.samsung-soc/20668 This patch set is made based on Kukjin Kim for-next branch. Andrew Bresticker (1): ARM: dts: exynos5420: add i2s controllers Padmavathi Venna (7): platform: Increase platform name size ASoC: Samsung: I2S: Add quirks as driver data in I2S ARM: dts: Change i2s compatible string on exynos5250 ASoC: Samsung: I2S: Modify driver to give more flexibility ASoC: Samsung: I2S: Modify the I2S driver to support I2S on Exynos5420 ARM: dts: wm8994: Add wm8994 support on smdk5420 ARM: dts: Enable sound support on smdk5420 .../devicetree/bindings/sound/samsung-i2s.txt | 25 ++- arch/arm/boot/dts/exynos5250.dtsi | 9 +- arch/arm/boot/dts/exynos5420-smdk5420.dts | 60 ++++++ arch/arm/boot/dts/exynos5420.dtsi | 44 +++++ include/linux/mod_devicetable.h | 2 +- include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s-regs.h | 51 ++++-- sound/soc/samsung/i2s.c | 205 +++++++++++++++----- 8 files changed, 312 insertions(+), 85 deletions(-) -- 1.7.4.4

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Re: [alsa-devel] [PATCH v3 1/4] ASoc: kirkwood: simplify probe error
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:17:39AM +0200, Jean-Francois Moine wrote: > The function kirkwood_i2s_dev_remove() may be used when probe fails. Looking at this deeper, I'm not happy with this. > +static int kirkwood_i2s_dev_remove(struct platform_device *pdev) > +{ > + struct kirkwood_dma_data *priv = dev_get_drvdata(&pdev->dev); > + > + snd_soc_unregister_component(&pdev->dev); ... > @@ -519,30 +532,17 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) > > err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, > soc_dai, 1); > + if (err) { > + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); > + goto fail; > + } > + return 0; > > +fail: > + kirkwood_i2s_dev_remove(pdev); What this means is that if snd_soc_register_component() fails, we end up calling snd_soc_unregister_component(). This may be fine with the way snd_soc_unregister_component() is currently implemented, but you're making the assumption that it's fine to call snd_soc_unregister_component() for a device which hasn't been registered. Technically, this is a layering violation, which makes this change fragile if the behaviour of snd_soc_unregister_component() changes in the future. For the sake of two calls in the error path, I don't think the benefits of this patch outweigh the risk.

2 1

Re: [alsa-devel] [PATCH v3 1/4] ASoc: kirkwood: simplify probe error
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:17:39AM +0200, Jean-Francois Moine wrote: > @@ -519,30 +532,17 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) > > err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, > soc_dai, 1); > - if (!err) > - return 0; > - dev_err(&pdev->dev, "snd_soc_register_component failed\n"); > - > - if (!IS_ERR(priv->extclk)) > - clk_disable_unprepare(priv->extclk); > - clk_disable_unprepare(priv->clk); > + if (err) { > + dev_err(&pdev->dev, "snd_soc_register_component failed\n"); > + goto fail; > + } > + return 0; This doesn't apply to mainline as there's a clk_put() there.

3 3

Re: [alsa-devel] [PATCH v3 3/4] ASoc: kirkwood: add DT support
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:18:32AM +0200, Jean-Francois Moine wrote: > The kirkwood audio driver is used without DT in the Kirkwood machine. > This patch adds a DT compatible definition for use in other Marvell > machines such as the Armada 88AP510 (Dove). > > Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> > --- > sound/soc/kirkwood/kirkwood-i2s.c | 27 ++++++++++++++++------ > 1 file changed, 20 insertions(+), 7 deletions(-) This misses documentation for the DT conversion. Documentation should be created at the same time as a driver is converted. I think you should combine patches 3 and 4 together given that you add the documentation and change the DT compatible name you've given this driver in this patch in patch 4. > > diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c > index a13fffb..234b5a1 100644 > --- a/sound/soc/kirkwood/kirkwood-i2s.c > +++ b/sound/soc/kirkwood/kirkwood-i2s.c > @@ -12,7 +12,6 @@ > > #include <linux/init.h> > #include <linux/module.h> > -#include <linux/platform_device.h> Why are you getting rid of this include? It still makes use of platform devices and the like, so it _should_ continue to include this. Don't rely on implicit includes.

1 0

Re: [alsa-devel] [PATCH v3 2/4] ASoc: kirkwood: merge kirkwood-i2c and kirkwood-dma
by Russell King - ARM Linux 03 Aug '13

03 Aug '13

On Wed, Jul 31, 2013 at 08:18:06AM +0200, Jean-Francois Moine wrote: > To avoid the declaration of a 'kirkwood-pcm-audio' device in the DT, > this patch merges the kirkwood-i2c and kirkwood-dma drivers into one > module associated with 'kirkwood-i2s'. This is broken unless you use my patch. You can't just stick the two drivers together because they both make use of the platform device's driver data field for two incompatible uses. I've already explained that, and I've even sent you a patch to fix that, but you seem to ignore anything that you didn't write. So... I'm going to work on this issue this weekend and merge the two drivers myself, and send Mark my own patches to do this.

1 1

[alsa-devel] [PATCH v2 7/9] ALSA: usb-audio: remove is_playback from implicit feedback quirks
by Eldad Zack 03 Aug '13

03 Aug '13

An implicit feedback endpoint can only be a capture source. The consumer (sink) of the implicit feedback endpoint is therefore limited to playback EPs. Check if the target endpoint is a playback first and remove redundant checks. Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/pcm.c | 40 +++++++++++++++++++--------------------- 1 file changed, 19 insertions(+), 21 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bb2e0f5..af30e08 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -334,41 +334,39 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, { struct usb_host_interface *alts; struct usb_interface *iface; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; int implicit_fb = 0; unsigned int ep; + /* Implicit feedback sync EPs consumers are always playback EPs */ + if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 3); + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 3); - if (!iface || iface->num_altsetting == 0) - return -EINVAL; + if (!iface || iface->num_altsetting == 0) + return -EINVAL; - alts = &iface->altsetting[1]; - goto add_sync_ep; - } + alts = &iface->altsetting[1]; + goto add_sync_ep; break; case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ case USB_ID(0x0763, 0x2081): - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 2); + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 2); - if (!iface || iface->num_altsetting == 0) - return -EINVAL; + if (!iface || iface->num_altsetting == 0) + return -EINVAL; - alts = &iface->altsetting[1]; - goto add_sync_ep; - } + alts = &iface->altsetting[1]; + goto add_sync_ep; } - if (is_playback && - attr == USB_ENDPOINT_SYNC_ASYNC && + if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && altsd->bInterfaceProtocol == 2 && altsd->bNumEndpoints == 1 && -- 1.8.1.5

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[alsa-devel] pcm1792a codec driver
by Michael Trimarchi 03 Aug '13

03 Aug '13

Hi Mark Yesterday I have done the pcm1792a codec driver and tested connected to the pandaboard es. The driver is working using spi/i2s connection with the codec and support l/r attenuation and digital mute. Other fmt are supported but not tested. Anyway it's a base of the complete driver. Hope that I can take again the hardware for feature release. Can I submit this initial support patch of it? Michael

1 0

[alsa-devel] end of playback ... design question
by Knut Petersen 02 Aug '13

02 Aug '13

Hi everybody! Although I looked at quite a bit of source code and documentation I have to admit that I am confused about one question: How should alsa playback be stopped in a way that _guarantees_ that no artifacts remaining in the drivers ringbuffer are played back again? Let´s assume we play stereo 16bit audio data using the rme96 driver, period size set to 0x800 frames. At the end of the penultimate frame playing will not be stopped even if there is only one frame of audio data remaining. Obviously that means 0x7ff frames of already played data is played again after that one valid frame, plus possibly a few that are played during the short time from interrupt request to the real stop of the playback hardware (about 5 frames on an old Pentium-M 1.2GHz). Now that could be fixed in different ways: A: Always ensure that there is a bit of silence (at least hardware buffer size plus a few frames) at the end of all audio data. Easy for the user , but only a workaround. B: At the rme96 driver level that condition can be detected and the relevant part of the buffer can be filled with zeros. I did that in an experimental change. It does work, but it´s definitely only a hack as it fixes the problem for only one device. C: The application playing audio data could pad all audio data to the period size with zeros. That would prevent playback of that max 0x7ff frames of garbage, but the few samples that are played during the time from interrupt request to the execution of the stop trigger do escape D: Don´t try to drain the buffer but pad with zeros and set a stop threshold. Workaround for applications should also work with older kernels. E: The alsa core could detect that condition and solve it equivalent to solution D I think that´s the proper solution. Did I miss something or do you agree? cu, Knut

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Re: [alsa-devel] HBR audio with intel HDA
by Anssi Hannula 02 Aug '13

02 Aug '13

02.08.2013 19:14, Ashecrow kirjoitti: > > Le 2 août 2013 à 15:42, Anssi Hannula a écrit : > >> 02.08.2013 16:11, Ashecrow kirjoitti: >>> Hi Anssi, >>> Thanks for your answer >>> >>> Test on kernel 3.8 + non-pcm bit in AES0: >>> aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif >>> => same result: no signal >> >> Hmh, OK. >> >>> With kernel upgrade to 3.9: same result >>> >>> >>> Note that I have tested with alsa* 1.025 and intel-hda driver included in kernel >>> >>> Do you have any other idea? >> >> You could try DEV=1 instead of DEV=0 (according to your log DEV=0 is >> correct, though), and truehd (thd.spdif from ffmpeg samples). >> >> If neither of those work, could you provide another alsa-info while >> playing back using AES0=0x06 (with alsa debug enabled again)? >> > I have already tried DEV=1 but it does the same. Moreover, let me note that I play DTS and dolby fine with DEV=0 in xbmc. So I guess it is the right device. OK. Note that in some cases you can hear stereo audio or AC-3/DTS even when using wrong DEV=x (but HBR or multichannel audio would not work). > see attached alsa-info.txt_dev0_kernel-3.8_aes0_dts-hd which is result of playing dts-hd spdif encapsulated with kernel 3.8 in debug mode and AES0=0x06 through command bellow > > aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif > > see attached alsa-info.txt_dev0_kernel-3.8_aes0_true-hd which is result of playing dts-hd spdif encapsulated with kernel 3.8 in debug mode and AES0=0x06 through command bellow > > aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 thd.spdif > Looks like your kernel is definitely too old to have working HBR on Intel (the "ASP channel x => slot y" messages should have x=y), so you'll need 3.9+. Unfortunately I don't see anything else wrong (and HBR+intel works for me), so I don't have anything else to suggest at the moment. Sorry. > Thanks for your help > >> >> >>> Thanks >>> >>> >>> Le 1 août 2013 à 18:37, Anssi Hannula a écrit : >>> >>>> 01.08.2013 19:29, ashecrow42(a)free.fr kirjoitti: >>>>> Hi there, >>>> >>>> Hi! >>>> >>>>> I have troubles using HBR audio (DTS-HD and Dolby TrueHD). With every >>>>> test, I get no signal in the AV receiver. I have tested using a dts-hd >>>>> encapsulated in spdif sample file (found there >>>>> http://samples.mplayerhq.hu/hdmi/) and commands bellow >>>>> >>>>> - aplay -D hdmi devices >>>>> - aplay -D hw devices >>>>> - aplay -D plughw devices >>>> >>>> You need to set the non-pcm bit in AES0. Use e.g. "-D hdmi:AES0=0x06". >>>> >>>>> Note that I use ubuntu raring with 3.8 kernel that includes Xingchao's >>>>> HBR kernel patches. >>>>> >>>>> >>>>> A. Configurations I have tested: >>>>> >>>>> >>>>> - hda driver upgrade (alsa devel ubuntu team daily builds) >>>>> >>>>> - upgrade alsa-lib & alsa-utils (alsa-testing launchpad) >>>>> >>>>> - kernel upgrade to 3.9 >>>> >>>> IIRC 3.9 (or newer) may indeed be needed since you have both hdmi and spdif. >>>> >>>>> - snd_hda_codec_realtek module backlist (to try to disable SPDIF) >>>>> /etc/modprobe.d/blacklist.conf >>>>> >>>>> - snd_hda_codec_realtek index = -2 in /etc/modprobe.d/alsa-base.conf >>>>> >>>>> - fresh new ubuntu raring install with no modifications at all: still >>>>> no signal >>>>> >>>>> B. Info: >>>>> >>>>> cat /proc/asound/card0/codec* | grep Codec >>>>> Codec: Realtek ALC892 >>>>> Codec: Intel CougarPoint HDMI >>>>> >>>>> I use zotac ZBOX ID82 (intel HM65 chipset with hdmi and spdif) >>>>> >>>>> with alsa debug and alsa debug verbose enabled in kernel I got the >>>>> alsa-info.txt attached file produced on raring with no modification >>>>> while playing dts-hd sample file with command: >>>>> >>>>> >>>>> aplay -D hdmi:CARD=PCH,DEV=0 -c8 -fs16_le -r192000 dts.spdif >>>> >>>> As per above you need to use >>>> aplay -D hdmi:CARD=PCH,DEV=0,AES0=0x06 -c8 -fs16_le -r192000 dts.spdif >>>> >>>>> >>>>> C. I add that: >>>>> >>>>> - In every test I have non HD audio protocols (DTS and dolby) working >>>>> well in the same configuration >>>>> >>>>> - I have tried with windows just to see if my hardware is broken and I >>>>> get DTS-HD audio with no problem >>>>> >>>>> - Takashi, I've read your HD-Audio.txt file and as far as I see, I have >>>>> no known error message in kernel log >>>>> >>>>> >>>>> => I and Wingchao Wang have conclued that it is driver or codec >>>>> related. Can you help me please? >>>>> >>>>> Thanks a lot >>>>> PS: sorry if you receive this email twice, I have sent it again as I >>>>> was not member of the list the first time. >>>> >>>> >>>> -- >>>> Anssi Hannula >>> >> >> >> -- >> Anssi Hannula > -- Anssi Hannula

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[alsa-devel] [PATCH] ASoC: dapm: Fix empty list check in dapm_new_mux()
by Lars-Peter Clausen 02 Aug '13

02 Aug '13

list_first_entry() will always return a valid pointer, even if the list is empty. So the check whether path is NULL will always be false. So we end up calling dapm_create_or_share_mixmux_kcontrol() with a path struct that points right in the middle of the widget struct and by trying to modify the path the widgets memory will become corrupted. Fix this by using list_emtpy() to check if the widget doesn't have any paths. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This is the real fix for the rt5640 crash. --- sound/soc/soc-dapm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5f64c16..196cea7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -747,13 +747,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - if (!path) { + if (list_empty(&w->sources)) { dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); return -EINVAL; } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); if (ret < 0) return ret; -- 1.8.0

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[alsa-devel] HBR audio with intel HDA
by ashecrow42＠free.fr 02 Aug '13

02 Aug '13

Hi there, I have troubles using HBR audio (DTS-HD and Dolby TrueHD). With every test, I get no signal in the AV receiver. I have tested using a dts-hd encapsulated in spdif sample file (found there http://samples.mplayerhq.hu/hdmi/) and commands bellow - aplay -D hdmi devices - aplay -D hw devices - aplay -D plughw devices Note that I use ubuntu raring with 3.8 kernel that includes Xingchao's HBR kernel patches. A. Configurations I have tested: - hda driver upgrade (alsa devel ubuntu team daily builds) - upgrade alsa-lib & alsa-utils (alsa-testing launchpad) - kernel upgrade to 3.9 - snd_hda_codec_realtek module backlist (to try to disable SPDIF) /etc/modprobe.d/blacklist.conf - snd_hda_codec_realtek index = -2 in /etc/modprobe.d/alsa-base.conf - fresh new ubuntu raring install with no modifications at all: still no signal B. Info: cat /proc/asound/card0/codec* | grep Codec Codec: Realtek ALC892 Codec: Intel CougarPoint HDMI I use zotac ZBOX ID82 (intel HM65 chipset with hdmi and spdif) with alsa debug and alsa debug verbose enabled in kernel I got the alsa-info.txt attached file produced on raring with no modification while playing dts-hd sample file with command: aplay -D hdmi:CARD=PCH,DEV=0 -c8 -fs16_le -r192000 dts.spdif C. I add that: - In every test I have non HD audio protocols (DTS and dolby) working well in the same configuration - I have tried with windows just to see if my hardware is broken and I get DTS-HD audio with no problem - Takashi, I've read your HD-Audio.txt file and as far as I see, I have no known error message in kernel log => I and Wingchao Wang have conclued that it is driver or codec related. Can you help me please? Thanks a lot PS: sorry if you receive this email twice, I have sent it again as I was not member of the list the first time.

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[alsa-devel] [PATCH 01/10] ASoC: dapm: Move snd_soc_dapm_update from dapm context to card
by Lars-Peter Clausen 02 Aug '13

02 Aug '13

The update field of a DAPM context is only assigned while the card's dapm_mutex is locked, the field is also cleared again while the mutex is stil locked. So there will only ever be one DAPM context at a time with a non-NULL update field. So it is safe to move the update field from the DAPM context struct to the card struct. Doing so will allow further cleanups in this area. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc-dapm.h | 2 -- include/sound/soc.h | 1 + sound/soc/soc-dapm.c | 22 +++++++++++----------- 3 files changed, 12 insertions(+), 13 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e77c6f5..3397292 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -575,8 +575,6 @@ struct snd_soc_dapm_context { struct delayed_work delayed_work; unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - struct snd_soc_dapm_update *update; - void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); diff --git a/include/sound/soc.h b/include/sound/soc.h index 6eabee7..b1e1f96 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1042,6 +1042,7 @@ struct snd_soc_card { /* Generic DAPM context for the card */ struct snd_soc_dapm_context dapm; struct snd_soc_dapm_stats dapm_stats; + struct snd_soc_dapm_update *update; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_card_root; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 366daef..7449e27 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1425,7 +1425,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, static void dapm_widget_update(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_update *update = dapm->update; + struct snd_soc_dapm_update *update = dapm->card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; unsigned int wi; @@ -1959,9 +1959,9 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mux_update_power(dapm, kcontrol, mux, e); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2002,9 +2002,9 @@ int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_context *dapm, int ret; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm->update = update; + card->update = update; ret = soc_dapm_mixer_update_power(dapm, kcontrol, connect); - dapm->update = NULL; + card->update = NULL; mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); @@ -2693,11 +2693,11 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2775,11 +2775,11 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); @@ -2928,11 +2928,11 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.reg = e->reg; update.mask = mask; update.val = val; - widget->dapm->update = &update; + card->update = &update; soc_dapm_mux_update_power(widget->dapm, kcontrol, mux, e); - widget->dapm->update = NULL; + card->update = NULL; } mutex_unlock(&card->dapm_mutex); -- 1.8.0

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[alsa-devel] [PATCH] ALSA: hda - Fix missing fixup for Mac Mini with STAC9221
by Takashi Iwai 02 Aug '13

02 Aug '13

A fixup for Apple Mac Mini was lost during the adaption to the generic parser because the fallback for the generic ID 8384:7680 was dropped, and it resulted in the silence output (and maybe other problems). Unfortunately, just adding the missing subsystem ID wasn't enough, in this case. The subsystem ID of this machine is 0000:0100 (what Apple thought...?), and since snd_hda_pick_fixup() doesn't take the vendor id zero into account, the driver ignored this entry. Now it's fixed to regard the vendor id zero as a valid value. Reported-and-tested-by: Linus Torvalds <torvalds(a)linux-foundation.org> Cc: <stable(a)vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/patch_sigmatel.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46..48a9d00 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } if (id < 0 && quirk) { - for (q = quirk; q->subvendor; q++) { + for (q = quirk; q->subvendor || q->subdevice; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 92b9b43..6d1924c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2819,6 +2819,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = { /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */ static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = { + SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3), SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1), SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2), SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2), -- 1.8.3.1

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[alsa-devel] [PATCH 1/2] ASoC: dapm: Check return value of snd_soc_cnew()
by Lars-Peter Clausen 01 Aug '13

01 Aug '13

snd_soc_cnew() can return NULL, so we should check the result before trying to use it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d74c356..b4fae87 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -671,8 +671,10 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name, prefix); - kcontrol->private_free = dapm_kcontrol_free; kfree(long_name); + if (!kcontrol) + return -ENOMEM; + kcontrol->private_free = dapm_kcontrol_free; ret = dapm_kcontrol_data_alloc(w, kcontrol); if (ret) { -- 1.8.0

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