February 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: cht_bsw_rt5645: Fix writing to string literal
by Carlo Caione 03 Mar '16

03 Mar '16

From: Carlo Caione <carlo(a)endlessm.com> We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2d3afdd..a7b96a9 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -367,8 +367,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) + if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00")) + card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL); + snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 2.5.0

5 6

Re: [alsa-devel] Compute Stick baytrail audio problem
by Giacomo Comes 03 Mar '16

03 Mar '16

On Tue, Feb 23, 2016 at 12:32:23PM +0900, Toyo Abe wrote: > > Hi, > > I'd struggled a similar underrun problem on my baytrail device. The dmesg showed > the following warnings. > > [15356.757938] had: Driver detected 2 missed buffer done interrupt(s)!!!! > [15356.761594] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.774775] had: Unable to clear UNDERRUN bits > [15356.791312] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.808148] had: Unable to clear UNDERRUN bits > [15356.824692] had: Driver detected 2 missed buffer done interrupt(s)!!!! > > After some investigation, I found some problems on the driver and fixed a few of > them. I pushed the fix to my github. I hope it might help you. > https://raw.githubusercontent.com/toyoabe/baytrailaudio/fix-underrun/hdmi_a… > > Best Regards, > -toyo Hi, that was it!. I added your changes and now the audio works. I'm happy. I got finally every subsystem of the Compute Stick working. I have a minor question tought. Alsamixer says: This sound device does not have any controls. Is there a way to change the volume without enabling pulseaudio? Just for documentation, here is the small change that Toyo did to make the underrun disappear: diff -Nraub a/drivers/gpu/drm/i915/i915_irq.c b/drivers/gpu/drm/i915/i915_irq.c --- a/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:42.406016576 -0400 +++ b/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:58.714016576 -0400 @@ -2717,11 +2717,14 @@ int pipe = 1; spin_lock_irqsave(&dev_priv->irq_lock, irqflags); + i915_enable_lpe_pipestat(dev_priv, pipe); + imr = I915_READ(VLV_IMR); /* Audio is on Stream A */ imr &= ~I915_LPE_PIPE_A_INTERRUPT; I915_WRITE(VLV_IMR, imr); - i915_enable_lpe_pipestat(dev_priv, pipe); + I915_WRITE(VLV_IER, ~imr); + POSTING_READ(VLV_IER); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); return 0; @@ -2769,7 +2772,10 @@ spin_lock_irqsave(&dev_priv->irq_lock, irqflags); imr = I915_READ(VLV_IMR); imr |= I915_LPE_PIPE_A_INTERRUPT; + I915_WRITE(VLV_IER, ~imr); I915_WRITE(VLV_IMR, imr); + POSTING_READ(VLV_IMR); + i915_disable_lpe_pipestat(dev_priv, pipe); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); diff -Nraub a/sound/hdmi_audio/intel_mid_hdmi_audio.c b/sound/hdmi_audio/intel_mid_hdmi_audio.c --- a/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:42.414016576 -0400 +++ b/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:58.722016576 -0400 @@ -1128,7 +1128,6 @@ pr_debug("HDMI status =0x%x\n", hdmi_status); if (hdmi_status & AUD_CONFIG_MASK_UNDERRUN) { i++; - hdmi_status &= ~AUD_CONFIG_MASK_UNDERRUN; had_write_register(AUD_HDMI_STATUS_v2, hdmi_status); } else break; Thanks again. Giacomo > > Pierre-Louis Bossart wrote: > > On 02/19/2016 02:39 PM, Giacomo Comes wrote: > >> Hi, > >> I have recently bought a Intel Compute Stick and I am having trouble with= > > the audio. > >> Intel provides a custom Ubuntu release that support audio, but I want to = > > run the > >> linux distribution of my choice on it (openSUSE 13.2 for the moment). > >> > >> In order to have HDMI audio support it is necessary to compile a kernel w= > > ith this patch > >> (https://raw.githubusercontent.com/01org/baytrailaudio/master/hdmi_audio_= > > 20150319.patch) > >> provided by Intel for kernel 3.16.7 (https://01.org/ubuntu-hdmi) > >> > >> I compiled the openSUSE kernel using the default openSUSE .config plus th= > > e following: > >> CONFIG_SUPPORT_HDMI=3Dy > >> CONFIG_SND_SOC=3Dm > >> CONFIG_SND_SOC_INTEL_SST=3Dm > >> CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=3Dm > >> > >> I booted the new kernel and aplay -l gives this output: > >> **** List of PLAYBACK Hardware Devices **** > >> card 0: IntelHDMI [IntelHDMI], device 0: IntelHDMI [IntelHDMI] > >> Subdevices: 0/1 > >> Subdevice #0: subdevice #0 > >> > >> but when I run: > >> aplay /usr/share/sounds/alsa/test.wav > >> the audio plays for .2 seconds and then nothing more. > >> Sometime it will play later a little bit more and stop again. > >> At some point these messages appears: > >> underrun!!! (at least 29325.946 ms long) > >> underrun!!! (at least 1535.889 ms long) > >> underrun!!! (at least 3196.213 ms long) > >> > >> > >> I have looked around on google for a hint on how to fix this issue but I = > > found nothing. > >> Any idea where to look for a solution? > >> > >> Attached there are the dmesg and lsmod output for ubuntu > >> with the working audio and openSUSE 13.2 plus patch > >> without the working audio. > >> I can provide any other information you need. > > > > There are enough messages that make me believe something is very wrong > > from the start > > > > 19.664486] had: ******** HAD DRIVER loading.. Ver: 0.01.003 > > [ 19.664549] hdmi_audio_probe dma_mask: 0 > > [ 19.664937] card->dev is NULL!!!!! Should not be this case > > [ 19.668176] [drm] mid_hdmi_audio_register: Scheduling HDMI audio work qu= > > eue > > [ 19.670745] [drm:i915_had_wq] *ERROR* Checking for HDMI connection at bo= > > ot > > > > Some of us at Intel are trying to clean this up but it'll take time. > > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

3 4

[alsa-devel] feeding back pkgsrc changs
by Thomas Klausner 03 Mar '16

03 Mar '16

Hi! I've just updated the pkgsrc packages for alsa-lib, alsa-utils and alsa-plugins to 1.1.0 and noticed that there are quite a number of local patches we have. I'd like to get them integrated. When I visited the alsa-projects web page, I noticed that the link to the bug tracker is dead. What is the recommended method to provide the patches? Thanks, Thomas

3 29

[alsa-devel] [PATCH 0/5] ASoC: pxa: Fix enum ctl accesses
by Takashi Iwai 02 Mar '16

02 Mar '16

Hi, here is a patchset to correct the wrong type accesses in pxa drivers for enum ctls. I didn't put Cc to stable for them, supposing these are only for 32bit systems. If not, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (5): ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type ASoC: pxa: magician: Fix enum ctl accesses in a wrong type ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type sound/soc/pxa/corgi.c | 12 ++++++------ sound/soc/pxa/magician.c | 6 +++--- sound/soc/pxa/poodle.c | 12 ++++++------ sound/soc/pxa/spitz.c | 12 ++++++------ sound/soc/pxa/tosa.c | 12 ++++++------ 5 files changed, 27 insertions(+), 27 deletions(-) -- 2.7.2

3 11

[alsa-devel] [PATCH v4 22/22] sound/usb: Use Media Controller API to share media resources
by Shuah Khan 02 Mar '16

02 Mar '16

Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- Changes since v3: - Fixed Kconfig to handle the following 1. CONFIG_MEDIA_SUPPORT and CONFIG_SND_USB_AUDIO are both modules CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 2. CONFIG_MEDIA_SUPPORT=y and CONFIG_SND_USB_AUDIO=m CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 3. CONFIG_MEDIA_SUPPORT=y and CONFIG_SND_USB_AUDIO=y CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 4. CONFIG_MEDIA_SUPPORT=m and CONFIG_SND_USB_AUDIO=y This is when we don't want CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER selected sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 +++ sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 72 +++++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 11 files changed, 448 insertions(+), 5 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index a452ad7..d14bf41 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,6 +15,7 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE + select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -22,6 +23,9 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_AUDIO_USE_MEDIA_CONTROLLER + bool + config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2d2d122..8dca3c4 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,6 +15,8 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o +snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o + snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 1f09d95..35fe256 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,6 +66,7 @@ #include "format.h" #include "power.h" #include "stream.h" +#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -561,6 +562,11 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; + if (quirk->media_device) { + /* don't want to fail when media_device_create() fails */ + media_device_create(chip, intf); + } + usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -617,6 +623,14 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } + /* + * Nice to check quirk && quirk->media_device + * need some special handlings. Doesn't look like + * we have access to quirk here + * Acceses mixer_list + */ + media_device_delete(chip); + /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca..34a0898 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,6 +105,8 @@ struct snd_usb_endpoint { struct list_head list; }; +struct media_ctl; + struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -156,6 +158,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c new file mode 100644 index 0000000..cff1459 --- /dev/null +++ b/sound/usb/media.c @@ -0,0 +1,318 @@ +/* + * media.c - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ + +#include <linux/init.h> +#include <linux/list.h> +#include <linux/mutex.h> +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/pcm.h> +#include <sound/core.h> + +#include "usbaudio.h" +#include "card.h" +#include "mixer.h" +#include "media.h" + +static int media_enable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->enable_source) + return mctl->media_dev->enable_source(&mctl->media_entity, + &mctl->media_pipe); + return 0; +} + +static void media_disable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->disable_source) + mctl->media_dev->disable_source(&mctl->media_entity); +} + +int media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream) +{ + struct media_device *mdev; + struct media_ctl *mctl; + struct device *pcm_dev = &pcm->streams[stream].dev; + u32 intf_type; + int ret = 0; + u16 mixer_pad; + struct media_entity *entity; + + mdev = subs->stream->chip->media_dev; + if (!mdev) + return -ENODEV; + + if (subs->media_ctl) + return 0; + + /* allocate media_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; + mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; + mixer_pad = 1; + } else { + intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; + mctl->media_pad.flags = MEDIA_PAD_FL_SINK; + mixer_pad = 2; + } + mctl->media_entity.name = pcm->name; + media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) + goto err1; + + mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, + MAJOR(pcm_dev->devt), + MINOR(pcm_dev->devt)); + if (!mctl->intf_devnode) { + ret = -ENOMEM; + goto err2; + } + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &mctl->intf_devnode->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + ret = -ENOMEM; + goto err3; + } + + /* create link between mixer and audio */ + media_device_for_each_entity(entity, mdev) { + switch (entity->function) { + case MEDIA_ENT_F_AUDIO_MIXER: + ret = media_create_pad_link(entity, mixer_pad, + &mctl->media_entity, 0, + MEDIA_LNK_FL_ENABLED); + if (ret) + goto err4; + break; + } + } + + subs->media_ctl = mctl; + return 0; + +err4: + media_remove_intf_link(mctl->intf_link); +err3: + media_devnode_remove(mctl->intf_devnode); +err2: + media_device_unregister_entity(&mctl->media_entity); +err1: + kfree(mctl); + return ret; +} + +void media_stream_delete(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl && mctl->media_dev) { + struct media_device *mdev; + + mdev = subs->stream->chip->media_dev; + if (mdev && media_devnode_is_registered(&mdev->devnode)) { + media_devnode_remove(mctl->intf_devnode); + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + subs->media_ctl = NULL; + } +} + +int media_start_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + return media_enable_source(mctl); + return 0; +} + +void media_stop_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + media_disable_source(mctl); +} + +int media_mixer_init(struct snd_usb_audio *chip) +{ + struct device *ctl_dev = &chip->card->ctl_dev; + struct media_intf_devnode *ctl_intf; + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + struct media_mixer_ctl *mctl; + u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; + int ret; + + if (!mdev) + return -ENODEV; + + ctl_intf = chip->ctl_intf_media_devnode; + if (!ctl_intf) { + ctl_intf = media_devnode_create(mdev, intf_type, 0, + MAJOR(ctl_dev->devt), + MINOR(ctl_dev->devt)); + if (!ctl_intf) + return -ENOMEM; + chip->ctl_intf_media_devnode = ctl_intf; + } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + + if (mixer->media_mixer_ctl) + continue; + + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; + mctl->media_entity.name = chip->card->mixername; + mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; + mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; + mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, + mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) { + kfree(mctl); + return ret; + } + + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &ctl_intf->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + kfree(mctl); + return -ENOMEM; + } + mctl->intf_devnode = ctl_intf; + mixer->media_mixer_ctl = mctl; + } + return 0; +} + +static void media_mixer_delete(struct snd_usb_audio *chip) +{ + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + + if (!mdev) + return; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + struct media_mixer_ctl *mctl; + + mctl = mixer->media_mixer_ctl; + if (!mixer->media_mixer_ctl) + continue; + + if (media_devnode_is_registered(&mdev->devnode)) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + mixer->media_mixer_ctl = NULL; + } + if (media_devnode_is_registered(&mdev->devnode)) + media_devnode_remove(chip->ctl_intf_media_devnode); + chip->ctl_intf_media_devnode = NULL; +} + +int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) +{ + struct media_device *mdev; + struct usb_device *usbdev = interface_to_usbdev(iface); + int ret; + + mdev = media_device_get_devres(&usbdev->dev); + if (!mdev) + return -ENOMEM; + if (!mdev->dev) { + /* register media device */ + mdev->dev = &usbdev->dev; + if (usbdev->product) + strlcpy(mdev->model, usbdev->product, + sizeof(mdev->model)); + if (usbdev->serial) + strlcpy(mdev->serial, usbdev->serial, + sizeof(mdev->serial)); + strcpy(mdev->bus_info, usbdev->devpath); + mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); + media_device_init(mdev); + } + if (!media_devnode_is_registered(&mdev->devnode)) { + ret = media_device_register(mdev); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't register media device. Error: %d\n", + ret); + return ret; + } + } + + /* save media device - avoid lookups */ + chip->media_dev = mdev; + + /* Create media entities for mixer and control dev */ + ret = media_mixer_init(chip); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't create media mixer entities. Error: %d\n", + ret); + + /* clear saved media_dev */ + chip->media_dev = NULL; + + return ret; + } + return 0; +} + +void media_device_delete(struct snd_usb_audio *chip) +{ + struct media_device *mdev = chip->media_dev; + + media_mixer_delete(chip); + + if (mdev) { + if (media_devnode_is_registered(&mdev->devnode)) + media_device_unregister(mdev); + chip->media_dev = NULL; + } +} diff --git a/sound/usb/media.h b/sound/usb/media.h new file mode 100644 index 0000000..f66b4d7 --- /dev/null +++ b/sound/usb/media.h @@ -0,0 +1,72 @@ +/* + * media.h - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ +#ifndef __MEDIA_H + +#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER + +#include <media/media-device.h> +#include <media/media-entity.h> +#include <sound/asound.h> + +struct media_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad; + struct media_pipeline media_pipe; +}; + +/* + * One source pad each for SNDRV_PCM_STREAM_CAPTURE and + * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link + * to AUDIO Source +*/ +#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) + +struct media_mixer_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; + struct media_pipeline media_pipe; +}; + +int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface); +void media_device_delete(struct snd_usb_audio *chip); +int media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream); +void media_stream_delete(struct snd_usb_substream *subs); +int media_start_pipeline(struct snd_usb_substream *subs); +void media_stop_pipeline(struct snd_usb_substream *subs); +#else +static inline int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) + { return 0; } +static inline void media_device_delete(struct snd_usb_audio *chip) { } +static inline int media_stream_init(struct snd_usb_substream *subs, + struct snd_pcm *pcm, int stream) + { return 0; } +static inline void media_stream_delete(struct snd_usb_substream *subs) { } +static inline int media_start_pipeline(struct snd_usb_substream *subs) + { return 0; } +static inline void media_stop_pipeline(struct snd_usb_substream *subs) { } +#endif +#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3417ef3..f378944 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,6 +3,8 @@ #include <sound/info.h> +struct media_mixer_ctl; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -22,6 +24,7 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; + struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9245f52..c3b8486 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,6 +35,7 @@ #include "pcm.h" #include "clock.h" #include "power.h" +#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -715,10 +716,14 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; + ret = media_start_pipeline(subs); + if (ret) + return ret; + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto err_ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -732,22 +737,27 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - return -EINVAL; + ret = -EINVAL; + goto err_ret; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; + +err_ret: + media_stop_pipeline(subs); + return ret; } /* @@ -759,6 +769,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; + media_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1219,6 +1230,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; + int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1232,7 +1244,12 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - return setup_hw_info(runtime, subs); + ret = setup_hw_info(runtime, subs); + if (ret == 0) + ret = media_stream_init(subs, as->pcm, direction); + if (ret) + snd_usb_autosuspend(subs->stream->chip); + return ret; } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1241,6 +1258,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); + media_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..9d087b1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,6 +2886,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4dc577..6c7f5c1 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,6 +36,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "media.h" /* * free a substream @@ -52,6 +53,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + media_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85..a161c7c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,6 +30,9 @@ * */ +struct media_device; +struct media_intf_devnode; + struct snd_usb_audio { int index; struct usb_device *dev; @@ -60,6 +63,8 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ + struct media_device *media_dev; + struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -110,6 +115,7 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; + bool media_device; const void *data; }; -- 2.5.0

3 4

[alsa-devel] [PATCH] ASoC: intel: remove unused variable
by Sudip Mukherjee 02 Mar '16

02 Mar '16

The variable cmd_id is only assigned some value and is never used. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/intel/atom/sst/sst_ipc.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3dc7358..8afa6fe 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -318,7 +318,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, union ipc_header_high msg_high; u32 msg_low; struct ipc_dsp_hdr *dsp_hdr; - unsigned int cmd_id; msg_high = msg->mrfld_header.p.header_high; msg_low = msg->mrfld_header.p.header_low_payload; @@ -357,7 +356,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, return; /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; - cmd_id = dsp_hdr->cmd_id; dev_dbg(sst_drv_ctx->dev, "cmd_id %d\n", dsp_hdr->cmd_id); if (sst_wake_up_block(sst_drv_ctx, msg_high.part.result, msg_high.part.drv_id, -- 1.9.1

4 13

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: distinguish the different SSC
by Songjun Wu 02 Mar '16

02 Mar '16

Cpu_dai id always equals 0, can't distinguish the different SSC. Use platform_device id to record and distinguish the different SSC. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- drivers/misc/atmel-ssc.c | 1 + sound/soc/atmel/atmel_ssc_dai.c | 27 ++++++++++++++++++--------- 2 files changed, 19 insertions(+), 9 deletions(-) diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index e11a0bd..0516ecd 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -34,6 +34,7 @@ struct ssc_device *ssc_request(unsigned int ssc_num) if (ssc->pdev->dev.of_node) { if (of_alias_get_id(ssc->pdev->dev.of_node, "ssc") == ssc_num) { + ssc->pdev->id = ssc_num; ssc_valid = 1; break; } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ba8def5..2768970 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -285,7 +285,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; int ret; @@ -346,7 +347,8 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; @@ -392,7 +394,8 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; ssc_p->daifmt = fmt; return 0; @@ -404,7 +407,8 @@ static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { - struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; switch (div_id) { case ATMEL_SSC_CMR_DIV: @@ -445,7 +449,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int id = dai->id; + struct platform_device *pdev = to_platform_device(dai->dev); + int id = pdev->id; struct atmel_ssc_info *ssc_p = &ssc_info[id]; struct ssc_device *ssc = ssc_p->ssc; struct atmel_pcm_dma_params *dma_params; @@ -772,7 +777,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, static int atmel_ssc_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir; @@ -795,7 +801,8 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, static int atmel_ssc_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir; @@ -824,11 +831,12 @@ static int atmel_ssc_trigger(struct snd_pcm_substream *substream, static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) { struct atmel_ssc_info *ssc_p; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); if (!cpu_dai->active) return 0; - ssc_p = &ssc_info[cpu_dai->id]; + ssc_p = &ssc_info[pdev->id]; /* Save the status register before disabling transmit and receive */ ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); @@ -852,12 +860,13 @@ static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) { struct atmel_ssc_info *ssc_p; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); u32 cr; if (!cpu_dai->active) return 0; - ssc_p = &ssc_info[cpu_dai->id]; + ssc_p = &ssc_info[pdev->id]; /* restore SSC register settings */ ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); -- 1.7.9.5

4 4

[alsa-devel] [PATCH v2] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 02 Mar '16

02 Mar '16

This is the initial imx-wm8958 device-tree-only machine driver working with fsl_sai driver. This sound card has three dai link, HIFI, VOICE and BT dai. HIFI dai link will support codec master and slave mode. VOICE an BT dai link have dummy cpu dai, and just support codec master mode. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/imx-audio-wm8958.txt | 49 +++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 432 +++++++++++++++++++++ 5 files changed, 498 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..50bad12 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,49 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - cpu-dai : The phandle of an CPU DAI controller + + - audio-codec : The phandle of the WM8958 audio codec + + - 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 could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + + - fsl,hifi-dai-master : If present, hifi dai works as master, and will + provide bit clock and frame clock. Otherwise, + hifi dai works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8960-audio"; + cpu-dai = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; + + fsl,hifi-dai-master; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..24eb66f --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,432 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link *dai_link; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_hifi_dai_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static u32 imx_wm8958_rates[] = { 8000, 16000, 32000, 48000 }; + +static struct snd_pcm_hw_constraint_list imx_wm8958_rate_constraints = { + .count = ARRAY_SIZE(imx_wm8958_rates), + .list = imx_wm8958_rates, +}; + +static int imx_wm8958_hifi_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + + if (!data->is_hifi_dai_master) + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_wm8958_rate_constraints); + + return ret; +} +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + /* + * Set GPIO1 pin function to reserve, so that DAC1 and ADC1 + * using shared LRCLK from DACLRCK1. + */ + snd_soc_update_bits(codec, WM8994_GPIO_1, 0x1f, 0x2); + + if (!data->is_hifi_dai_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (!data->is_hifi_dai_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } else if (id == VOICE_DAI) { + /* + * Set GPIO6 pin function to reserve, so that DAC2 and ADC2 + * using shared LRCLK from DACLRCK2. + */ + snd_soc_update_bits(codec, WM8994_GPIO_6, 0x1f, 0x2); + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && !data->is_hifi_dai_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static int imx_wm8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int ret, i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* need to enable mclk to write/read wm8958 register */ + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) { + ret = clk_prepare_enable(data->mclk[i]); + if (ret) + dev_warn(card->dev, + "Failed to enable MCLK%d: %d\n", + i + 1, ret); + } + } + } + break; + default: + break; + } + + return 0; +} + +static int imx_wm8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) + clk_disable_unprepare(data->mclk[i]); + } + break; + default: + break; + } + + card->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, + .startup = imx_wm8958_hifi_startup, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + cpu_np = of_parse_phandle(np, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + return -EINVAL; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find cpu dai platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->dai_link = imx_wm8958_dai_link; + + /* + * AIF1 support codec master and slave mode + * AIF2 and AIF3 just support codec master mode + */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,hifi-dai-master")) { + data->is_hifi_dai_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + data->is_hifi_dai_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (IS_ERR(data->mclk[i])) { + ret = PTR_ERR(data->mclk[i]); + dev_err(&pdev->dev, "failed to get mclk%d clock: %d\n", + i + 1, ret); + } else { + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + } + + data->dai_link[HIFI_DAI].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->dai_link[HIFI_DAI].platform_of_node = cpu_np; + + data->card.set_bias_level = imx_wm8958_set_bias_level; + data->card.set_bias_level_post = imx_wm8958_set_bias_level_post; + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = DAI_LINK_NUM; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: mediatek: Add machine driver for rt5650 rt5514 codec
by PC Liao 02 Mar '16

02 Mar '16

From: Koro Chen <koro.chen(a)mediatek.com> This is the DPCM based machine driver with rt5650 and rt5514. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../bindings/sound/mt8173-rt5650-rt5514.txt | 15 ++ sound/soc/mediatek/Kconfig | 28 ++- sound/soc/mediatek/Makefile | 3 +- sound/soc/mediatek/mt8173-rt5650-rt5514.c | 258 ++++++++++++++++++++ 4 files changed, 294 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt create mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5514.c diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt new file mode 100644 index 0000000..e8b3c80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt @@ -0,0 +1,15 @@ +MT8173 with RT5650 RT5514 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5514" +- mediatek,audio-codec: the phandles of rt5650 and rt5514 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5514"; + mediatek,audio-codec = <&rt5650 &rt5514>; + mediatek,platform = <&afe>; + }; + diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 6c88087..f7e789e 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -18,14 +18,25 @@ config SND_SOC_MT8173_MAX98090 If unsure select "N". config SND_SOC_MT8173_RT5650 - tristate "ASoC Audio driver for MT8173 with RT5650 codec" - depends on SND_SOC_MEDIATEK && I2C - select SND_SOC_RT5645 - help - This adds ASoC driver for Mediatek MT8173 boards - with the RT5650 audio codec. - Select Y if you have such device. - If unsure select "N". + tristate "ASoC Audio driver for MT8173 with RT5650 codec" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 audio codec. + Select Y if you have such device. + If unsure select "N". + +config SND_SOC_MT8173_RT5650_RT5514 + tristate "ASoC Audio driver for MT8173 with RT5650 RT5514 codecs" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + select SND_SOC_RT5514 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 and RT5514 codecs. + Select Y if you have such device. + If unsure select "N". config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" @@ -37,4 +48,3 @@ config SND_SOC_MT8173_RT5650_RT5676 with the RT5650 and RT5676 codecs. Select Y if you have such device. If unsure select "N". - diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile index bcdcb06..d486860 100644 --- a/sound/soc/mediatek/Makefile +++ b/sound/soc/mediatek/Makefile @@ -2,5 +2,6 @@ obj-$(CONFIG_SND_SOC_MEDIATEK) += mtk-afe-pcm.o # Machine support obj-$(CONFIG_SND_SOC_MT8173_MAX98090) += mt8173-max98090.o -obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o obj-$(CONFIG_SND_SOC_MT8173_RT5650) += mt8173-rt5650.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5514) += mt8173-rt5650-rt5514.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173-rt5650-rt5514.c new file mode 100644 index 0000000..58e0836 --- /dev/null +++ b/sound/soc/mediatek/mt8173-rt5650-rt5514.c @@ -0,0 +1,258 @@ +/* + * mt8173-rt5650-rt5514.c -- MT8173 machine driver with RT5650/5514 codecs + * + * Copyright (c) 2016 MediaTek Inc. + * Author: Koro Chen <koro.chen(a)mediatek.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 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" + +#define MCLK_FOR_CODECS 12288000 + +static const struct snd_soc_dapm_widget mt8173_rt5650_rt5514_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route mt8173_rt5650_rt5514_routes[] = { + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + {"Sub DMIC1L", NULL, "Int Mic"}, + {"Sub DMIC1R", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Headset Mic", NULL, "micbias1"}, + {"Headset Mic", NULL, "micbias2"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, +}; + +static const struct snd_kcontrol_new mt8173_rt5650_rt5514_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + + /* pll from mclk 12.288M */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, + params_rate(params) * 512); + if (ret) + return ret; + + /* sysclk from pll */ + ret = snd_soc_dai_set_sysclk(codec_dai, 1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + return 0; +} + +static struct snd_soc_ops mt8173_rt5650_rt5514_ops = { + .hw_params = mt8173_rt5650_rt5514_hw_params, +}; + +static struct snd_soc_jack mt8173_rt5650_rt5514_jack; + +static int mt8173_rt5650_rt5514_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + int ret; + + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* enable jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &mt8173_rt5650_rt5514_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "Can't new Headset Jack %d\n", ret); + return ret; + } + + return rt5645_set_jack_detect(codec, + &mt8173_rt5650_rt5514_jack, + &mt8173_rt5650_rt5514_jack, + &mt8173_rt5650_rt5514_jack); +} + +static struct snd_soc_dai_link_component mt8173_rt5650_rt5514_codecs[] = { + { + .dai_name = "rt5645-aif1", + }, + { + .dai_name = "rt5514-aif1", + }, +}; + +enum { + DAI_LINK_PLAYBACK, + DAI_LINK_CAPTURE, + DAI_LINK_CODEC_I2S, +}; + +/* Digital audio interface glue - connects codec <---> CPU */ +static struct snd_soc_dai_link mt8173_rt5650_rt5514_dais[] = { + /* Front End DAI links */ + [DAI_LINK_PLAYBACK] = { + .name = "rt5650_rt5514 Playback", + .stream_name = "rt5650_rt5514 Playback", + .cpu_dai_name = "DL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, + [DAI_LINK_CAPTURE] = { + .name = "rt5650_rt5514 Capture", + .stream_name = "rt5650_rt5514 Capture", + .cpu_dai_name = "VUL", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_capture = 1, + }, + /* Back End DAI links */ + [DAI_LINK_CODEC_I2S] = { + .name = "Codec", + .cpu_dai_name = "I2S", + .no_pcm = 1, + .codecs = mt8173_rt5650_rt5514_codecs, + .num_codecs = 2, + .init = mt8173_rt5650_rt5514_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &mt8173_rt5650_rt5514_ops, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +static struct snd_soc_codec_conf mt8173_rt5650_rt5514_codec_conf[] = { + { + .name_prefix = "Sub", + }, +}; + +static struct snd_soc_card mt8173_rt5650_rt5514_card = { + .name = "mtk-rt5650-rt5514", + .owner = THIS_MODULE, + .dai_link = mt8173_rt5650_rt5514_dais, + .num_links = ARRAY_SIZE(mt8173_rt5650_rt5514_dais), + .codec_conf = mt8173_rt5650_rt5514_codec_conf, + .num_configs = ARRAY_SIZE(mt8173_rt5650_rt5514_codec_conf), + .controls = mt8173_rt5650_rt5514_controls, + .num_controls = ARRAY_SIZE(mt8173_rt5650_rt5514_controls), + .dapm_widgets = mt8173_rt5650_rt5514_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_rt5514_widgets), + .dapm_routes = mt8173_rt5650_rt5514_routes, + .num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_rt5514_routes), +}; + +static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt8173_rt5650_rt5514_card; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5514_dais[i].platform_name) + continue; + mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node; + } + + mt8173_rt5650_rt5514_codecs[0].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); + if (!mt8173_rt5650_rt5514_codecs[0].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + mt8173_rt5650_rt5514_codecs[1].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); + if (!mt8173_rt5650_rt5514_codecs[1].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + mt8173_rt5650_rt5514_codec_conf[0].of_node = + mt8173_rt5650_rt5514_codecs[1].of_node; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + return ret; +} + +static const struct of_device_id mt8173_rt5650_rt5514_dt_match[] = { + { .compatible = "mediatek,mt8173-rt5650-rt5514", }, + { } +}; +MODULE_DEVICE_TABLE(of, mt8173_rt5650_rt5514_dt_match); + +static struct platform_driver mt8173_rt5650_rt5514_driver = { + .driver = { + .name = "mtk-rt5650-rt5514", + .of_match_table = mt8173_rt5650_rt5514_dt_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, + .probe = mt8173_rt5650_rt5514_dev_probe, +}; + +module_platform_driver(mt8173_rt5650_rt5514_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8173 RT5650 and RT5514 SoC machine driver"); +MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mtk-rt5650-rt5514"); + -- 1.7.9.5

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Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Liam Girdwood 02 Mar '16

02 Mar '16

On Tue, 2016-02-16 at 21:47 +0530, Jeeja KP wrote: > We have no mechanism to configure hw_params for DAIs which are hostless. > Like a loop or DAIs connected to internal sinks/sources. > Fwiw, a few years back Peter, Sebastien and I were considering a similar mechanism for the ABE on OMAP4. We never got around to implementing or upstreaming though.... One comment below. > Problem Statement: > During a BE-BE loop of DSP or a SRC->SINK path, we cannot program > hw_params. There are no hw_params from user as these are hostless. > This scenario can happens in below 2 cases: > > Case 1: BE-BE loop aka hostless stream through SoC DSP > Hostless stream can be defined by BE to BE connection and in this case > the path is enabled or disabled by the state of the DAPM graph. This > usually means there is a mixer control that can be used to connect or > disconnect the path between both DAIs. > Example: > |--------------/----------------| > | | > | | > codec out modem_in > | | > | | > ssp0_TX ssp1_RX > | | > | | > AIFI_Playback dummy_capture > > In the above case (SSP0,codec playback,SSp1 dai needs to be configured > and turned on with the selected hw_params). > > Case 2: SoC Internal Sink or Source > Example: > WOV SINK (DAPM Sink) > | > | > | > Switch > | > | > w1 > | > dmic rx (BE dai widget) > | > Capture(codec dai widget) > | > DMIC AIF > | > DMic > | > Soc DMic (MIC Widget) (SRC) > > In the above case BE (dmic RX) dai hw_params needs to applied. > > Approach: > 1. Allow params to be defined for other BE dai link also. this > currently applicable only for codec_codec link. > > 2. when params is defined for BE dai link, create params kcontrols > and associate it with the dai widgets. this will allow params to be > selected via Kcontrol for each dai defined in the dai link. > The only problem we have with using a kcontrol to define params is that we already have an existing kernel ioctl and public C API to define SW/HW params. This would add another mechanism, unless each new kcontrol allows modification of a single param ? Liam > 3. Before powering on the dai widget and if the dai is not powered > with the stream event, then dai_ops(hw_params) for the dai widget will > be called.This will apply the selected hw_params. > > 4. with the above, fix_up can be removed and can use .params to > select the parameter. > > Core Changes: > 1. params definition needs to changed, currently this is defined to used > only for codec codec link. define new flag to identifying codec > codec link. > > 2. if params is defined for any BE dai link, then create kcontrol for > each dai defined in the dai_link and associated the kcontrol to dai > widget. > > 3. In the powering sequence check if the dai widget and if the event is > not a stream event, SND_SOC_DAPM_STREAM_NOP , then call dai ops with > the selected params. This will apply the hw_params for dai. > > 1. params definition: > - Add new flag in struct snd_soc_dai_link. > - Based on this flag create, the codec codec link instead of > params > struct snd_soc_dai_link { > const struct snd_soc_pcm_stream *params; > unsigned int num_params; > > + /* flag to create codec codec based on the flag*/ > + unsigned int codec_link:1; > + > unsigned int dai_fmt; /* format to set on init */ > > enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ > > - creation of dai link based on the codec_link flag > static int soc_probe_link_dais(struct snd_soc_card *card, > int num, int order) > } > } else { > > - if (!dai_link->params) { > + if (!dai_link->codec_link) { > /* create the pcm */ > ret = soc_new_pcm(rtd, num); > if (ret < 0) { > > 2. when params is defined for BE dai link, create params kcontrols and > associate it with the dai widgets. this will allow params to be selected > via Kcontrol for each dai defined in the dai link. > a. Move the param control creation from snd_soc_dapm_new_pcm and add new > API to created kcontrols with the params. > +snd_soc_dapm_create_param_control(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > { > /* create kcontrol based on the params */ > } > > - modify the snd_soc_dapm_new_pcm to create the param control > int snd_soc_dapm_new_pcm(struct snd_soc_card *card, > const struct snd_soc_pcm_stream *params, > unsigned int num_params, > struct snd_soc_dapm_widget *source, > struct snd_soc_dapm_widget *sink) > { > > template.kcontrol_news = snd_soc_dapm_create_param_control(card, > params, num_params); > if (!template.kcontrol_news) { > dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", > link_name); > ret = -ENOMEM; > - goto outfree_private_value; > + goto outfree_link_name; > } > > > b. add new API snd_soc_dapm_new_dai_params() to create kcontrol for dai and associated it to dai. > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w) > +{ > + struct snd_kcontrol_new *kcontrol_news; > + > + kcontrol_news = snd_soc_dapm_create_param_control(card, params, > + num_params); > + if (!kcontrol_news) { > + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", > + dai_w->name); > + return -EINVAL; > + } > + > + dai_w->num_kcontrols = 1; > + dai_w->kcontrol_news = kcontrol_news; > + > + dai_w->params = params; > + dai_w->num_params = num_params; > + > + return 0; > + > +} > + > +++ b/include/sound/soc-dapm.h > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w); > > > c. if the be dai link has .params defined, call new API snd_soc_dapm_new_dai_params > > +static int soc_link_be_params_widget(struct snd_soc_card *card, > + struct snd_soc_dai *dai, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > +{ > + int ret; > + struct snd_soc_dapm_widget *w; > + > + if (!strcmp(dai->name, "snd-soc-dummy")) > + return 0; > + > + if (dai->playback_widget) { > + w = dai->playback_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + if (dai->capture_widget) { > + w = dai->capture_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + return 0; > +} > + > +static int soc_link_be_params_dai_widgets(struct snd_soc_card *card, > + struct snd_soc_dai_link *dai_link, > + struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + int ret, i; > + > + for (i = 0; i < rtd->num_codecs; i++) { > + struct snd_soc_dai *codec = rtd->codec_dais[i]; > + > + ret = soc_link_be_params_widget(card, codec, > + dai_link->params, > + dai_link->num_params); > + > + if (!ret) > + return ret; > + } > + > + return soc_link_be_params_widget(card, cpu_dai, dai_link->params, > + dai_link->num_params); > +} > + > > @@ -1651,6 +1714,13 @@ static int soc_probe_link_dais(struct snd_soc_card *card, > dai_link->stream_name, ret); > return ret; > } > + if (dai_link->params && dai_link->no_pcm) { > + /* create dai link controls */ > + ret = soc_link_be_params_dai_widgets(card, > + dai_link, rtd); > + if (ret) > + return ret; > + } > } else { > INIT_DELAYED_WORK(&rtd->delayed_work, > codec2codec_close_delayed_work); > > > 3. Applying the hw_params for dai widget > > --- a/sound/soc/soc-dapm.c > +++ b/sound/soc/soc-dapm.c > @@ -1398,9 +1398,64 @@ static void dapm_seq_check_event(struct snd_soc_card * > card, > } > } > > +static int snd_soc_dai_params(struct snd_soc_dapm_widget *w, int event) > +{ > + struct snd_soc_dai *dai; > + const struct snd_soc_pcm_stream *config = w->params + w->params_select; > + struct snd_pcm_substream substream; > + struct snd_pcm_hw_params *params = NULL; > + u64 fmt; > + int ret = 0; > + > + if (WARN_ON(!config)) > + return -EINVAL; > + > + dai = w->priv; > + > + if (config->formats) { > + fmt = ffs(config->formats) - 1; > + } else { > + dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n", > + config->formats); > + fmt = 0; > + } > + > + /* Currently very limited parameter selection */ > + params = kzalloc(sizeof(*params), GFP_KERNEL); > + if (!params) { > + ret = -ENOMEM; > + goto out; > + } > + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = > + config->rate_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = > + config->rate_max; > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min > + = config->channels_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max > + = config->channels_max; > + > + memset(&substream, 0, sizeof(substream)); > + if (w->id == snd_soc_dapm_dai_in) > + substream.stream = SNDRV_PCM_STREAM_CAPTURE; > + else > + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; > + > + ret = soc_dai_hw_params(&substream, params, dai); > + if (ret < 0) > + goto out; > +out: > + kfree(params); > + return ret; > +} > + > /* Apply the coalesced changes from a DAPM sequence */ > static void dapm_seq_run_coalesced(struct snd_soc_card *card, > - struct list_head *pending) > + struct list_head *pending, > + int event) > { > struct snd_soc_dapm_context *dapm; > struct snd_soc_dapm_widget *w; > @@ -1413,6 +1468,14 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm = w->dapm; > > list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_PRE_PMU); > + } > + } > + > + list_for_each_entry(w, pending, power_list) { > WARN_ON(reg != w->reg || dapm != w->dapm); > w->power = w->new_power; > > @@ -1447,6 +1510,13 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU); > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD); > } > + list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_POST_PMD); > + } > + } > } > > /* Apply a DAPM power sequence. > @@ -1482,7 +1552,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > if (sort[w->id] != cur_sort || w->reg != cur_reg || > w->dapm != cur_dapm || w->subseq != cur_subseq) { > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > @@ -1545,7 +1615,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > } > > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > -- > 1.7.9.5 > > Comments welcome, based on discussion we will send patches. > > --Jeeja

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