June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ALSA: hda/hdmi: Add Intel silent stream support
by Harsha Priya 01 Jul '20

01 Jul '20

External HDMI receivers have analog circuitry that needs to be powered-on when exiting standby, and a mechanism to detect PCM v. IEC61937 data. These two steps take time and up to 2-3 seconds of audio may be muted when starting playback. Intel hardware can keep the link active with a 'silent stream', so that the receiver does not go through those two steps when valid audio is transmitted. This mechanism relies on an info packet and preventing the codec from going to D3, which will increase the platform static power consumption. The info packet assumes a basic 2ch stereo, and the silent stream is enabled when connecting a monitor. In case of format changes the detection of PCM v. IEC61937 needs to be re-run. In this case there is no way to avoid the 2-3s mute. The silent stream is enabled with a Kconfig option, as well as a kernel parameter should there be a need to override the build time default. This approach is used based on the power_save capability as an example, but in the future, it may be used with a kcontrol, depending on UCM support for HDaudio legacy. Silent stream is supported in Intel platforms Skylake and beyond. Tested HDMI plug-out plug-in from Intel Cometlake based Chromebook connected to few different monitors. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- Change History: v2: - Kconfig symbol depends on SND_HDA_INTEL. - Added valid eld check to along side of silent stream enable check, under which silent stream is enabled. - Error prints added for codec power up, power down functions. - The silent stream is enabled with a Kconfig/module parameter following the power_save capability as an example. But in the future, it may be used with a kcontrol, depending on UCM support for HDaudio legacy. v1: - Initial version. sound/pci/hda/Kconfig | 16 ++++++++++++++ sound/pci/hda/patch_hdmi.c | 45 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 61 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 7ba542e45a3d..1c4758d58fc3 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -232,4 +232,20 @@ config SND_HDA_POWER_SAVE_DEFAULT endif +config SND_HDA_INTEL_HDMI_SILENT_STREAM + bool "Enable Silent Stream always for HDMI" + depends on SND_HDA_INTEL + help + Intel hardware has a feature called 'silent stream', that + keeps external HDMI receiver's analog circuitry powered on + avoiding 2-3 sec silence during playback start. This mechanism + relies on an info packet and preventing the codec from going to + D3. (increasing the platform static power consumption when a + HDMI receiver is plugged-in). 2-3 sec silence at the playback + start is expected whenever there is format change. (default is + 2 channel format). + Say Y to enable Silent Stream feature. + +endif + endmenu diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index fbd7cc6026d8..68b81ea176c6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -42,6 +42,11 @@ static bool enable_acomp = true; module_param(enable_acomp, bool, 0444); MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)"); +static bool enable_silent_stream = +IS_ENABLED(CONFIG_SND_HDA_INTEL_HDMI_SILENT_STREAM); +module_param(enable_silent_stream, bool, 0644); +MODULE_PARM_DESC(enable_silent_stream, "Enable Silent Stream for HDMI devices"); + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -1634,21 +1639,61 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, snd_hda_power_down_pm(codec); } +static void silent_stream_enable(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + codec_dbg(codec, "hdmi: enabling silent stream for NID %d\n", + per_pin->pin_nid); + + mutex_lock(&per_pin->lock); + if (!per_pin->channels) + per_pin->channels = 2; + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + mutex_unlock(&per_pin->lock); +} + /* update ELD and jack state via audio component */ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; + bool monitor_prev, monitor_next; mutex_lock(&per_pin->lock); eld->monitor_present = false; + monitor_prev = per_pin->sink_eld.monitor_present; eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, per_pin->dev_id, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); eld->eld_valid = (eld->eld_size > 0); update_eld(codec, per_pin, eld, 0); + monitor_next = per_pin->sink_eld.monitor_present; mutex_unlock(&per_pin->lock); + + /* + * Power-up will call hdmi_present_sense, so the PM calls + * have to be done without mutex held. + */ + + if (enable_silent_stream && eld->eld_valid) { + int pm_ret; + + if (!monitor_prev && monitor_next && eld->eld_valid) { + pm_ret = snd_hda_power_up_pm(codec); + if (pm_ret < 0) + codec_err(codec, + "Monitor plugged-in, Failed to power up codec ret=[%d]\n", + pm_ret); + silent_stream_enable(codec, per_pin); + } else if (monitor_prev && !monitor_next && !eld->eld_valid) { + pm_ret = snd_hda_power_down_pm(codec); + if (pm_ret < 0) + codec_err(codec, + "Monitor plugged-out, Failed to power down codec ret=[%d]\n", + pm_ret); + } + } } static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) -- 2.17.1

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[PATCH 0/5] soundwire: handle stream at the dailink level
by Bard Liao 01 Jul '20

01 Jul '20

Currently, stream is handled at the dai level. But we have to handle stream at the dailink level in the multi-cpu dailink usage. Pierre-Louis Bossart (5): soundwire: intel: implement get_sdw_stream() operations soundwire: stream: add helper to startup/shutdown streams soundwire: intel: remove stream allocation/free soundwire: cadence: allocate/free dma_data in set_sdw_stream soundwire: intel: don't free dma_data in DAI shutdown drivers/soundwire/cadence_master.c | 52 ++++++++++++----- drivers/soundwire/intel.c | 82 +++++++-------------------- drivers/soundwire/stream.c | 90 ++++++++++++++++++++++++++++++ include/linux/soundwire/sdw.h | 2 + 4 files changed, 149 insertions(+), 77 deletions(-) -- 2.17.1

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[GIT PULL] ASoC fixes for v5.8-rc3
by Mark Brown 30 Jun '20

30 Jun '20

The following changes since commit f141a422159a199f4c8dedb7e0df55b3b2cf16cd: ASoC: rockchip: Fix a reference count leak. (2020-06-18 17:21:58 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.8-rc3 for you to fetch changes up to bc5c7f55f5ea91e137fc7939435ed2e2bb6e5a15: AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module (2020-06-25 15:06:56 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.8 A few small driver specific fixes, nothing particularly dramatic. ---------------------------------------------------------------- Jaroslav Kysela (1): AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module Shengjiu Wang (2): ASoC: fsl_mqs: Don't check clock is NULL before calling clk API ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable Shuming Fan (1): ASoC: rt5682: fix the pop noise while OMTP type headset plugin sound/soc/amd/renoir/Makefile | 7 ++++--- sound/soc/codecs/rt5682.c | 9 ++++++++- sound/soc/fsl/fsl_mqs.c | 23 ++++++++++++++--------- 3 files changed, 26 insertions(+), 13 deletions(-)

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[PATCH] ALSA: usb-audio: Fix some typos
by Alexander Tsoy 30 Jun '20

30 Jun '20

Fix the following typos in comments and in the code: - KHz -> kHz - procssed -> processed Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/mixer_s1810c.c | 6 +++--- sound/usb/mixer_scarlett_gen2.c | 4 ++-- sound/usb/pcm.c | 4 ++-- sound/usb/quirks.c | 14 +++++++------- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 6483e47bafd0..c53a9773f310 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -554,11 +554,11 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) dev_info(&dev->dev, "Presonus Studio 1810c, device_setup: %u\n", chip->setup); if (chip->setup == 1) - dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + dev_info(&dev->dev, "(8out/18in @ 48kHz)\n"); else if (chip->setup == 2) - dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + dev_info(&dev->dev, "(6out/8in @ 192kHz)\n"); else - dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + dev_info(&dev->dev, "(8out/14in @ 96kHz)\n"); ret = snd_s1810c_init_mixer_maps(chip); if (ret < 0) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 74c00c905d24..0ffff7640892 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -332,7 +332,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_SPDIF] = { .id = 0x180, - /* S/PDIF outputs aren't available at 192KHz + /* S/PDIF outputs aren't available at 192kHz * but are included in the USB mux I/O * assignment message anyway */ @@ -401,7 +401,7 @@ static const struct scarlett2_device_info s18i20_gen2_info = { .dst_descr = "Analogue Output %02d Playback" }, [SCARLETT2_PORT_TYPE_SPDIF] = { - /* S/PDIF outputs aren't available at 192KHz + /* S/PDIF outputs aren't available at 192kHz * but are included in the USB mux I/O * assignment message anyway */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a777d36c4f5a..e42241087b77 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1694,8 +1694,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs, int processed = urb->transfer_buffer_length / ep->stride; int est_delay; - /* ignore the delay accounting when procssed=0 is given, i.e. - * silent payloads are procssed before handling the actual data + /* ignore the delay accounting when processed=0 is given, i.e. + * silent payloads are processed before handling the actual data */ if (!processed) return; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fca72730a802..c551141f337e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1144,14 +1144,14 @@ static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) #define MAUDIO_SET 0x01 /* parse device_setup */ #define MAUDIO_SET_COMPATIBLE 0x80 /* use only "win-compatible" interfaces */ #define MAUDIO_SET_DTS 0x02 /* enable DTS Digital Output */ -#define MAUDIO_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */ +#define MAUDIO_SET_96K 0x04 /* 48-96kHz rate if set, 8-48kHz otherwise */ #define MAUDIO_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */ #define MAUDIO_SET_DI 0x10 /* enable Digital Input */ #define MAUDIO_SET_MASK 0x1f /* bit mask for setup value */ -#define MAUDIO_SET_24B_48K_DI 0x19 /* 24bits+48KHz+Digital Input */ -#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48KHz+No Digital Input */ -#define MAUDIO_SET_16B_48K_DI 0x11 /* 16bits+48KHz+Digital Input */ -#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48KHz+No Digital Input */ +#define MAUDIO_SET_24B_48K_DI 0x19 /* 24bits+48kHz+Digital Input */ +#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48kHz+No Digital Input */ +#define MAUDIO_SET_16B_48K_DI 0x11 /* 16bits+48kHz+Digital Input */ +#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48kHz+No Digital Input */ static int quattro_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1597,7 +1597,7 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) /* * M-Audio Fast Track C400/C600 - when packets are not skipped, real - * world latency varies by approx. +/- 50 frames (at 96KHz) each time + * world latency varies by approx. +/- 50 frames (at 96kHz) each time * the stream is (re)started. When skipping packets 16 at endpoint * start up, the real world latency is stable within +/- 1 frame (also * across power cycles). @@ -1839,7 +1839,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* * MaxPacketsOnly attribute is erroneously set in endpoint * descriptors. As a result this card produces noise with - * all sample rates other than 96 KHz. + * all sample rates other than 96 kHz. */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; -- 2.26.2

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[PATCH 1/2] ALSA: usb-audio: Fix packet size calculation
by Alexander Tsoy 30 Jun '20

30 Jun '20

Commit f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") introduced a regression for devices which have playback endpoints with bInterval > 1. Fix this by taking ep->datainterval into account. Note that frame and fps are actually mean packet and packets per second in the code introduces by the mentioned commit. This will be fixed in a follow-up patch. Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208353 Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/endpoint.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 9bea7d3f99f8..11f23778f0a5 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1093,6 +1093,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->freqn = get_usb_high_speed_rate(rate); ep->fps = 8000; } + ep->fps >>= ep->datainterval; ep->sample_rem = rate % ep->fps; ep->framesize[0] = rate / ep->fps; -- 2.26.2

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[PATCH] ALSA: core: Warn on empty module
by Takashi Iwai 30 Jun '20

30 Jun '20

The module argument passed to snd_card_new() must be a valid non-NULL pointer when the module support is enabled. Since ASoC driver passes the argument from each snd_soc_card definition, one may forget to set the owner field and lead to a NULL module easily. For catching such an overlook, add a WARN_ON() in snd_card_new(). Also, put the card->module assignment in the ifdef block for a very minor optimization. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/init.c b/sound/core/init.c index b02a99766351..0478847ba2b8 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -203,7 +203,10 @@ int snd_card_new(struct device *parent, int idx, const char *xid, mutex_unlock(&snd_card_mutex); card->dev = parent; card->number = idx; +#ifdef MODULE + WARN_ON(!module); card->module = module; +#endif INIT_LIST_HEAD(&card->devices); init_rwsem(&card->controls_rwsem); rwlock_init(&card->ctl_files_rwlock); -- 2.16.4

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[PATCH 0/4] soundwire: add SoundWire 1.2 spec support
by Bard Liao 30 Jun '20

30 Jun '20

This series adds basic support for SoundWire 1.2. Pierre-Louis Bossart (4): soundwire: add definitions for 1.2 spec soundwire: bus_type: convert open-coded while() to for() loop soundwire: extend SDW_SLAVE_ENTRY soundwire: bus: initialize bus clock base and scale registers drivers/soundwire/bus.c | 107 ++++++++++++++++++++++ drivers/soundwire/bus_type.c | 19 ++-- include/linux/mod_devicetable.h | 2 + include/linux/soundwire/sdw.h | 11 ++- include/linux/soundwire/sdw_registers.h | 117 +++++++++++++++++++++++- scripts/mod/devicetable-offsets.c | 2 + scripts/mod/file2alias.c | 6 +- 7 files changed, 250 insertions(+), 14 deletions(-) -- 2.17.1

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Re: XHCI vs PCM2903B/PCM2904 part 2
by Mathias Nyman 30 Jun '20

30 Jun '20

On 30.6.2020 16.08, Rik van Riel wrote: > I misread the code, it's not a bitfield, so state 1 means an endpoint marked with running state. The next urb is never getting a response, though. > > However, the xhci spec says an endpoint is halted upon a babble error. I was looking at the same, so according to specs this state shouldn't be possible. > > The code right above the babble handling case adds halted into the endpoint state itself. Does the code handling the babble error need to do something similar to trigger cleanup elsewhere? It's a flag to prevent ringing the doorbell for a halted endpoint. Anyway, reset endpoint is meant to recover an endpoint in a halted state. Resetting non-halted endpoints will just lead to a context state error, and besides, isoc endpoints shouldn't halt. Anyways, I haven't got any better idea at the moment. You can try and see what a forced reset does with: diff --git a/drivers/usb/host/xhci-ring.c b/drivers/usb/host/xhci-ring.c index 2c255d0620b0..d79aca0df6d4 100644 --- a/drivers/usb/host/xhci-ring.c +++ b/drivers/usb/host/xhci-ring.c @@ -1900,8 +1900,7 @@ static int xhci_requires_manual_halt_cleanup(struct xhci_hcd *xhci, * endpoint anyway. Check if a babble halted the * endpoint. */ - if (GET_EP_CTX_STATE(ep_ctx) == EP_STATE_HALTED) - return 1; + return 1; return 0; } Traces also showed thet endpoint doorbell was rang after th babble error, so we know that didn't help restarting the endpoint. -Mathias

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[PATCH 1/2] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 30 Jun '20

30 Jun '20

Just convert Rockchip rk3328 audio codec binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 --------- .../bindings/sound/rockchip,rk3328-codec.yaml | 61 +++++++++++++++++++ 2 files changed, 61 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..08a74e28e483 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + + clock-names: + items: + - const: pclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + +examples: + - | + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODEC>; + clock-names = "pclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = 100; + }; -- 2.27.0

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[PATCH] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 30 Jun '20

30 Jun '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 58 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 56 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..fcafc8ecb131 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,59 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/** + * Select proper clock source for internal ratio mode + */ +static int fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + bool clk_sel[2]; + + rate[0] = in_rate; + rate[1] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + if (i == ASRC_CLK_MAP_LEN) { + select_clk[j] = OUTCLK_ASRCK1_CLK; + clk_sel[j] = false; + } else { + select_clk[j] = i; + clk_sel[j] = true; + } + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (!clk_sel[IN] || !clk_sel[OUT]) + select_clk[IN] = INCLK_NONE; + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +653,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +666,14 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + ret = fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + if (ret) { + dev_err(dai->dev, "fail to select clock\n"); + return ret; + } + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

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