December 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 00/13] ASoC: Intel: Skylake: Update to SKL driver
by Vinod Koul 12 Dec '15

12 Dec '15

This update bring in new APIs in core and usage of those in SKL driver. The series is broadly of three parts - first two are fixes on the driver - then we add DMA resume capability in HDA core and use it to save restore DMA link values and add DMA resume capability in driver - lastly, we add support for MISCBDCGE configuration in core and use it while resetting and also for phrase detection per HW recommendation This should be merged thru ASoC tree due to obvious dependencies. Takashi would need your ack on core patches changes in v2: - move msicbdge api to driver and as a consequence sqash into into first use - fix lkp warning Dharageswari.R (1): ASoC: Intel: Skylake: Use CGCTL.MISCBDCGE for Phrase detection notification Jayachandran B (2): ALSA: hdac: Increase timeout value for link power check ASoC: Intel: Skylake: fix reset controller sequencing Jeeja KP (8): ASoC: Intel: Skylake: Clear stream registers before stream setup ASoC: Intel: Skylake: Fix to set pipe state to invalid when deleting ALSA: hdac: Add support for hda DMA Resume capability ALSA: hdac: couple the hda DMA stream in cleanup ASoC: Intel: Skylake: enable interrupt as wake source in active suspend ASoC: Intel: Skylake: Add DMA resume position in Trigger resume/suspend ASoC: Intel: Skylake: Reconfigure Link stream on suspend/resume ASoC: Intel: Skylake: Add Resume capability in PCM info. Vinod Koul (2): ALSA: hdac: add snd_hdac_ext_bus_link_power_up_all ASoC: Intel: Skylake: manage link power in active suspend include/sound/hda_register.h | 9 +++++ include/sound/hdaudio_ext.h | 15 +++++++ sound/hda/ext/hdac_ext_controller.c | 29 ++++++++++++- sound/hda/ext/hdac_ext_stream.c | 72 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 2 + sound/soc/intel/skylake/skl-pcm.c | 56 ++++++++++++++++++++----- sound/soc/intel/skylake/skl-sst-cldma.c | 38 ++++++++++------- sound/soc/intel/skylake/skl-sst-ipc.c | 15 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 3 ++ sound/soc/intel/skylake/skl.c | 51 +++++++++++++++++++++-- sound/soc/intel/skylake/skl.h | 5 +++ 11 files changed, 265 insertions(+), 30 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH 05/14] ALSA: hdac: Add MISCBDCGE support
by Vinod Koul 12 Dec '15

12 Dec '15

On Sat, Dec 12, 2015 at 05:14:48AM +0800, kbuild test robot wrote: > Hi Jayachandran, > > [auto build test WARNING on asoc/for-next] > [also build test WARNING on next-20151211] > [cannot apply to sound/for-next v4.4-rc4] > > url: https://github.com/0day-ci/linux/commits/Vinod-Koul/ASoC-Intel-Skylake-Upda… > base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > config: mn10300-allmodconfig (attached as .config) > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # save the attached .config to linux build tree > make.cross ARCH=mn10300 > > All warnings (new ones prefixed by >>): > > sound/hda/ext/hdac_ext_controller.c: In function 'update_pci_dword': > >> sound/hda/ext/hdac_ext_controller.c:317:7: warning: 'data' is used uninitialized in this function [-Wuninitialized] > data &= ~mask; Thanks, will fix it in v2 -- ~Vinod

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[alsa-devel] [PATCH] ASoC: fsl_asrc: sound is wrong after suspend/resume
by Zidan Wang 12 Dec '15

12 Dec '15

The register ASRCFG is volatile, but some bits need to be recovered after suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_asrc.c | 6 ++++++ sound/soc/fsl/fsl_asrc.h | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 3e404ba..942e74bc 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -990,6 +990,9 @@ static int fsl_asrc_suspend(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); + regmap_read(asrc_priv->regmap, REG_ASRCFG, + &asrc_priv->regcache_cfg); + regcache_cache_only(asrc_priv->regmap, true); regcache_mark_dirty(asrc_priv->regmap); @@ -1010,6 +1013,9 @@ static int fsl_asrc_resume(struct device *dev) regcache_cache_only(asrc_priv->regmap, false); regcache_sync(asrc_priv->regmap); + regmap_update_bits(asrc_priv->regmap, REG_ASRCFG, + 0x1FFFC0, asrc_priv->regcache_cfg); + /* Restart enabled pairs */ regmap_update_bits(asrc_priv->regmap, REG_ASRCTR, ASRCTR_ASRCEi_ALL_MASK, asrctr); diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 68802cd..3695ca9 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -453,6 +453,8 @@ struct fsl_asrc { int asrc_rate; int asrc_width; + + u32 regcache_cfg; }; extern struct snd_soc_platform_driver fsl_asrc_platform; -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 12 Dec '15

12 Dec '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. This is a hardware bug/errata and reset is the only option. According to the reference manual, the software reset doesn't reset any control register but only internal hardware logics such as bit clock generator, status flags, and FIFO pointers. (Our purpose is just to reset the clock generator while the software reset is the only way to do that.) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. For asynchronous mode, TX will not be reset when RX is still running. In this case, i can't reproduce this issue. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1) refine the commit log and comments 2) Since slave mode doesn't use the clock generator, only apply the reset procedure to the master mode. sound/soc/fsl/fsl_sai.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 8832901..a55a7de 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -564,6 +564,17 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + if (!sai->is_slave_mode) { + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, + FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, + FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + } } break; default: -- 1.9.1

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[alsa-devel] Available frames > buffer size
by Rob Duncan 11 Dec '15

11 Dec '15

Is it ever correct for snd_pcm_avail() to return a value that’s greater than that returned by snd_pcm_hw_params_get_buffer_size()? I’m trying to track down some unexpected underruns in my application and I notice that when they occur I also see that the available frames count is wrong. For example here are a couple of lines from my logging: 2015-12-08T23:17:23.118817-08:00 X ? 2080 + 2080 2015-12-08T23:17:23.119524-08:00 X = 2080 ? 4126 The first log line: here I asked the device how many frames are available and it responds with 2080; I accordingly write 2080 frames. The second log line: the writei() call returns 2080. I immediately ask how many frames are available now (I usually get a small number in response ~ 20-30) but this time I get an XRUN. I call snd_pcm_recover() and then snd_pcm_avail() again and now I get 4126 frames. As you can see the time between these two logs is less than 1 millisecond, so I don’t understand how the PCM can go from 2080 frames available to >4096 frames available so quickly. Is this evidence of a bug in the device driver? My application is using a single channel dshare PCM (rt) that has an 8-channel device underneath. Here’s the configuration fragment: pcm_slave.cid_dsp { pcm "hw:0,0" channels 8 rate 44100 } pcm.rt_channel { type dshare ipc_key 1025 ipc_gid audio ipc_perm 0660 slave cid_dsp bindings { 0 5 } } pcm.rt { type plug slave.pcm rt_channel } The underlying hardware looks like this: access: MMAP_INTERLEAVED format: S16_LE subformat: STD channels: 8 rate: 44100 (44100/1) period_size: 1024 buffer_size: 4096 tstamp_mode: ENABLE period_step: 1 avail_min: 1 start_threshold: 1 stop_threshold: 1073741824 silence_threshold: 0 silence_size: 0 boundary: 1073741824 Thanks for any insight and help, Rob.

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[alsa-devel] [PATCH] ALSA: usb-audio: constify usb_protocol_ops structures
by Julia Lawall 11 Dec '15

11 Dec '15

The usb_protocol_ops structures are never modified, so declare them as const. Done with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/usb/midi.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index ee212e7..cc39f63 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -112,7 +112,7 @@ struct snd_usb_midi { struct usb_interface *iface; const struct snd_usb_audio_quirk *quirk; struct snd_rawmidi *rmidi; - struct usb_protocol_ops *usb_protocol_ops; + const struct usb_protocol_ops *usb_protocol_ops; struct list_head list; struct timer_list error_timer; spinlock_t disc_lock; @@ -671,31 +671,32 @@ static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint *ep, } } -static struct usb_protocol_ops snd_usbmidi_standard_ops = { +static const struct usb_protocol_ops snd_usbmidi_standard_ops = { .input = snd_usbmidi_standard_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_midiman_ops = { +static const struct usb_protocol_ops snd_usbmidi_midiman_ops = { .input = snd_usbmidi_midiman_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_midiman_packet, }; -static struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { +static const +struct usb_protocol_ops snd_usbmidi_maudio_broken_running_status_ops = { .input = snd_usbmidi_maudio_broken_running_status_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_cme_ops = { +static const struct usb_protocol_ops snd_usbmidi_cme_ops = { .input = snd_usbmidi_cme_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, }; -static struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { +static const struct usb_protocol_ops snd_usbmidi_ch345_broken_sysex_ops = { .input = ch345_broken_sysex_input, .output = snd_usbmidi_standard_output, .output_packet = snd_usbmidi_output_standard_packet, @@ -795,7 +796,7 @@ static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep, } } -static struct usb_protocol_ops snd_usbmidi_akai_ops = { +static const struct usb_protocol_ops snd_usbmidi_akai_ops = { .input = snd_usbmidi_akai_input, .output = snd_usbmidi_akai_output, }; @@ -835,7 +836,7 @@ static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = 2 + count; } -static struct usb_protocol_ops snd_usbmidi_novation_ops = { +static const struct usb_protocol_ops snd_usbmidi_novation_ops = { .input = snd_usbmidi_novation_input, .output = snd_usbmidi_novation_output, }; @@ -867,7 +868,7 @@ static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = count; } -static struct usb_protocol_ops snd_usbmidi_raw_ops = { +static const struct usb_protocol_ops snd_usbmidi_raw_ops = { .input = snd_usbmidi_raw_input, .output = snd_usbmidi_raw_output, }; @@ -883,7 +884,7 @@ static void snd_usbmidi_ftdi_input(struct snd_usb_midi_in_endpoint *ep, snd_usbmidi_input_data(ep, 0, buffer + 2, buffer_length - 2); } -static struct usb_protocol_ops snd_usbmidi_ftdi_ops = { +static const struct usb_protocol_ops snd_usbmidi_ftdi_ops = { .input = snd_usbmidi_ftdi_input, .output = snd_usbmidi_raw_output, }; @@ -927,7 +928,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = ep->max_transfer; } -static struct usb_protocol_ops snd_usbmidi_122l_ops = { +static const struct usb_protocol_ops snd_usbmidi_122l_ops = { .input = snd_usbmidi_us122l_input, .output = snd_usbmidi_us122l_output, }; @@ -1060,7 +1061,7 @@ static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint *ep, urb->transfer_buffer_length = ep->max_transfer - buf_free; } -static struct usb_protocol_ops snd_usbmidi_emagic_ops = { +static const struct usb_protocol_ops snd_usbmidi_emagic_ops = { .input = snd_usbmidi_emagic_input, .output = snd_usbmidi_emagic_output, .init_out_endpoint = snd_usbmidi_emagic_init_out,

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx
by Peter Ujfalusi 11 Dec '15

11 Dec '15

The condition for checking for XDAT being cleared was not correct. Fixes: 36bcecd0a73eb ("ASoC: davinci-mcasp: Correct TX start sequence") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c1c9c2e3525b..2ccb8bccc9d4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -223,8 +223,8 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp) /* wait for XDATA to be cleared */ cnt = 0; - while (!(mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & - ~XRDATA) && (cnt < 100000)) + while ((mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & XRDATA) && + (cnt < 100000)) cnt++; /* Release TX state machine */ -- 2.6.4

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Re: [alsa-devel] [Intel-gfx] [PATCH V4 1/2] dp/mst: add SDP stream support
by Libin Yang 11 Dec '15

11 Dec '15

Add Takashi and ALSA mailing list Best Regards, Libin On 12/02/2015 02:09 PM, libin.yang(a)linux.intel.com wrote: > From: Libin Yang <libin.yang(a)linux.intel.com> > > This adds code to initialise the SDP streams > for a sink in the simplest ordering. > > I've no idea how you'd want to control the > ordering at this level, so don't bother > until someone comes up with a use case. > > Reviewed-by: Ander Conselvan de Oliveira <conselvan2(a)gmail.com> > Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> > Signed-off-by: Dave Airlie <airlied(a)redhat.com> > --- > drivers/gpu/drm/drm_dp_mst_topology.c | 38 ++++++++++++++++++++++++++++++++--- > include/drm/drm_dp_mst_helper.h | 7 +++++-- > 2 files changed, 40 insertions(+), 5 deletions(-) > > diff --git a/drivers/gpu/drm/drm_dp_mst_topology.c b/drivers/gpu/drm/drm_dp_mst_topology.c > index 809959d..f50eb7b 100644 > --- a/drivers/gpu/drm/drm_dp_mst_topology.c > +++ b/drivers/gpu/drm/drm_dp_mst_topology.c > @@ -666,7 +666,9 @@ static int build_enum_path_resources(struct drm_dp_sideband_msg_tx *msg, int por > } > > static int build_allocate_payload(struct drm_dp_sideband_msg_tx *msg, int port_num, > - u8 vcpi, uint16_t pbn) > + u8 vcpi, uint16_t pbn, > + u8 number_sdp_streams, > + u8 *sdp_stream_sink) > { > struct drm_dp_sideband_msg_req_body req; > memset(&req, 0, sizeof(req)); > @@ -674,6 +676,9 @@ static int build_allocate_payload(struct drm_dp_sideband_msg_tx *msg, int port_n > req.u.allocate_payload.port_number = port_num; > req.u.allocate_payload.vcpi = vcpi; > req.u.allocate_payload.pbn = pbn; > + req.u.allocate_payload.number_sdp_streams = number_sdp_streams; > + memcpy(req.u.allocate_payload.sdp_stream_sink, sdp_stream_sink, > + number_sdp_streams); > drm_dp_encode_sideband_req(&req, msg); > msg->path_msg = true; > return 0; > @@ -1562,6 +1567,8 @@ static int drm_dp_payload_send_msg(struct drm_dp_mst_topology_mgr *mgr, > struct drm_dp_sideband_msg_tx *txmsg; > struct drm_dp_mst_branch *mstb; > int len, ret; > + u8 sinks[DRM_DP_MAX_SDP_STREAMS]; > + int i; > > mstb = drm_dp_get_validated_mstb_ref(mgr, port->parent); > if (!mstb) > @@ -1573,10 +1580,13 @@ static int drm_dp_payload_send_msg(struct drm_dp_mst_topology_mgr *mgr, > goto fail_put; > } > > + for (i = 0; i < port->num_sdp_streams; i++) > + sinks[i] = i; > + > txmsg->dst = mstb; > len = build_allocate_payload(txmsg, port->port_num, > id, > - pbn); > + pbn, port->num_sdp_streams, sinks); > > drm_dp_queue_down_tx(mgr, txmsg); > > @@ -2259,6 +2269,27 @@ out: > EXPORT_SYMBOL(drm_dp_mst_detect_port); > > /** > + * drm_dp_mst_port_has_audio() - Check whether port has audio capability or not > + * @mgr: manager for this port > + * @port: unverified pointer to a port. > + * > + * This returns whether the port supports audio or not. > + */ > +bool drm_dp_mst_port_has_audio(struct drm_dp_mst_topology_mgr *mgr, > + struct drm_dp_mst_port *port) > +{ > + bool ret = false; > + > + port = drm_dp_get_validated_port_ref(mgr, port); > + if (!port) > + return ret; > + ret = port->has_audio; > + drm_dp_put_port(port); > + return ret; > +} > +EXPORT_SYMBOL(drm_dp_mst_port_has_audio); > + > +/** > * drm_dp_mst_get_edid() - get EDID for an MST port > * @connector: toplevel connector to get EDID for > * @mgr: manager for this port > @@ -2283,6 +2314,7 @@ struct edid *drm_dp_mst_get_edid(struct drm_connector *connector, struct drm_dp_ > edid = drm_get_edid(connector, &port->aux.ddc); > drm_mode_connector_set_tile_property(connector); > } > + port->has_audio = drm_detect_monitor_audio(edid); > drm_dp_put_port(port); > return edid; > } > @@ -2566,7 +2598,7 @@ static void drm_dp_mst_dump_mstb(struct seq_file *m, > > seq_printf(m, "%smst: %p, %d\n", prefix, mstb, mstb->num_ports); > list_for_each_entry(port, &mstb->ports, next) { > - seq_printf(m, "%sport: %d: ddps: %d ldps: %d, %p, conn: %p\n", prefix, port->port_num, port->ddps, port->ldps, port, port->connector); > + seq_printf(m, "%sport: %d: ddps: %d ldps: %d, sdp: %d/%d, %p, conn: %p\n", prefix, port->port_num, port->ddps, port->ldps, port->num_sdp_streams, port->num_sdp_stream_sinks, port, port->connector); > if (port->mstb) > drm_dp_mst_dump_mstb(m, port->mstb); > } > diff --git a/include/drm/drm_dp_mst_helper.h b/include/drm/drm_dp_mst_helper.h > index 5340099..74b5888 100644 > --- a/include/drm/drm_dp_mst_helper.h > +++ b/include/drm/drm_dp_mst_helper.h > @@ -94,6 +94,7 @@ struct drm_dp_mst_port { > struct drm_dp_mst_topology_mgr *mgr; > > struct edid *cached_edid; /* for DP logical ports - make tiling work */ > + bool has_audio; > }; > > /** > @@ -215,13 +216,13 @@ struct drm_dp_sideband_msg_rx { > struct drm_dp_sideband_msg_hdr initial_hdr; > }; > > - > +#define DRM_DP_MAX_SDP_STREAMS 16 > struct drm_dp_allocate_payload { > u8 port_number; > u8 number_sdp_streams; > u8 vcpi; > u16 pbn; > - u8 sdp_stream_sink[8]; > + u8 sdp_stream_sink[DRM_DP_MAX_SDP_STREAMS]; > }; > > struct drm_dp_allocate_payload_ack_reply { > @@ -484,6 +485,8 @@ int drm_dp_mst_hpd_irq(struct drm_dp_mst_topology_mgr *mgr, u8 *esi, bool *handl > > enum drm_connector_status drm_dp_mst_detect_port(struct drm_connector *connector, struct drm_dp_mst_topology_mgr *mgr, struct drm_dp_mst_port *port); > > +bool drm_dp_mst_port_has_audio(struct drm_dp_mst_topology_mgr *mgr, > + struct drm_dp_mst_port *port); > struct edid *drm_dp_mst_get_edid(struct drm_connector *connector, struct drm_dp_mst_topology_mgr *mgr, struct drm_dp_mst_port *port); > > >

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[alsa-devel] [PATCH v5] sun4i-codec: add inputs
by Danny Milosavljevic 11 Dec '15

11 Dec '15

Hi, this is the fifth version of the patch that adds inputs to sun4i-codec. (Previous versions have been on the linux-sunxi mailing list only) The inputs added are: - FM-In Left and Right - Line-In Left and Right - Mic1-In - Mic2-In Changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. I successfully tested it on an A20 board using alsamixer, headphones, a radio and my ears. Note that because of missing capturing support I tested only the mixing, for Mic, Line, and FM. The patches are on top of <git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git>, branch "sunxi/for-next". Regards, Danny Danny (1): b/sound/soc/sunxi/sun4i-codec.c | 153 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 150 insertions(+), 3 deletions(-) Signed-off-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> --- diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da..106b568 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -57,9 +57,20 @@ #define SUN4I_CODEC_DAC_ACTL_DACAENR (31) #define SUN4I_CODEC_DAC_ACTL_DACAENL (30) #define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LNG (26) +#define SUN4I_CODEC_DAC_ACTL_FMG (23) +#define SUN4I_CODEC_DAC_ACTL_MICG (20) +#define SUN4I_CODEC_DAC_ACTL_LLNS (19) +#define SUN4I_CODEC_DAC_ACTL_RLNS (18) +#define SUN4I_CODEC_DAC_ACTL_LFMS (17) +#define SUN4I_CODEC_DAC_ACTL_RFMS (16) #define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) #define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) #define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_MIC1LS (12) +#define SUN4I_CODEC_DAC_ACTL_MIC1RS (11) +#define SUN4I_CODEC_DAC_ACTL_MIC2LS (10) +#define SUN4I_CODEC_DAC_ACTL_MIC2RS (9) #define SUN4I_CODEC_DAC_ACTL_DACPAS (8) #define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) #define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) @@ -84,8 +95,11 @@ #define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) #define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) #define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_PREG1_A10 (25) +#define SUN4I_CODEC_ADC_ACTL_PREG2_A10 (23) #define SUN4I_CODEC_ADC_ACTL_VADCG (20) #define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_LNRDF (16) #define SUN4I_CODEC_ADC_ACTL_PA_EN (4) #define SUN4I_CODEC_ADC_ACTL_DDE (3) #define SUN4I_CODEC_ADC_DEBUG (0x2c) @@ -94,7 +108,18 @@ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) #define SUN4I_CODEC_AC_SYS_VERI (0x38) -#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) + +/* MIC_PHONE_CAL register offsets and bit fields (A20 only) */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1 (29) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2 (26) +/* note: no idea where the output pins for the following are. */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTG (5) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTEN (4) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS3 (3) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS2 (2) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS1 (1) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS0 (0) struct sun4i_codec { struct device *dev; @@ -402,16 +427,94 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); +static DECLARE_TLV_DB_SCALE(sun4i_codec_linein_loopback_gain_scale, -150, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_fmin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_micin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_RANGE(sun7i_codec_micin_preamp_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0) +); +static DECLARE_TLV_DB_RANGE(sun4i_codec_micin_preamp_gain_scale_a10, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(3500, 300, 0) +); + +struct snd_kcontrol_new sun7ionly_codec_widgets[] = { + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), +}; -static const struct snd_kcontrol_new sun4i_codec_widgets[] = { +/* Note: this is modified by the sun4i_codec_probe() function. */ +static struct snd_kcontrol_new sun4i_codec_widgets[] = { SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, sun4i_codec_pa_volume_scale), + /* Line-In, FM-In, Mic1-In, Mic2-In */ + SOC_SINGLE_TLV("Line-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LNG, + 1, + 0, + sun4i_codec_linein_loopback_gain_scale), + SOC_SINGLE_TLV("FM-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_FMG, + 3, + 0, + sun4i_codec_fmin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MICG, + 7, + 0, + sun4i_codec_micin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic1-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1EN, + 1, + 0, + NULL), + SOC_SINGLE_TLV("Mic2-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2EN, + 1, + 0, + NULL), }; static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), + SOC_DAPM_SINGLE("Left Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LLNS, 1, 0), + SOC_DAPM_SINGLE("Left FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1LS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2LS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { @@ -419,6 +522,14 @@ static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Right Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RLNS, 1, 0), + SOC_DAPM_SINGLE("Right FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1RS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2RS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { @@ -462,8 +573,16 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), + + SND_SOC_DAPM_INPUT("Line-In Right"), + SND_SOC_DAPM_INPUT("Line-In Left"), + SND_SOC_DAPM_INPUT("FM-In Right"), + SND_SOC_DAPM_INPUT("FM-In Left"), + SND_SOC_DAPM_INPUT("Mic1-In"), + SND_SOC_DAPM_INPUT("Mic2-In"), }; +/* {sink, control, source} */ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { /* Left DAC Routes */ { "Left DAC", NULL, "DAC" }, @@ -490,6 +609,20 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Line-In, FM-In */ + { "Right Mixer", "Right Line-In Playback Switch", "Line-In Right" }, + { "Left Mixer", "Left Line-In Playback Switch", "Line-In Left" }, + { "Right Mixer", "Right FM-In Playback Switch", "FM-In Right" }, + { "Left Mixer", "Left FM-In Playback Switch", "FM-In Left" }, + + /* Mic1-In */ + { "Right Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + { "Left Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + + /* Mic2-In */ + { "Right Mixer", "Mic2-In Playback Switch", "Mic2-In" }, + { "Left Mixer", "Mic2-In Playback Switch", "Mic2-In" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { @@ -537,7 +670,7 @@ static const struct regmap_config sun4i_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, - .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, }; static const struct of_device_id sun4i_codec_of_match[] = { @@ -594,6 +727,20 @@ static int sun4i_codec_probe(struct platform_device *pdev) void __iomem *base; int ret; + if (of_device_is_compatible(pdev->dev.of_node, "allwinner,sun7i-a20-codec")) { + size_t i, j; + for (i = 0; i < ARRAY_SIZE(sun7ionly_codec_widgets); ++i) { + struct snd_kcontrol_new *new_kcontrol; + new_kcontrol = &sun7ionly_codec_widgets[i]; + for (j = 0; j < ARRAY_SIZE(sun4i_codec_widgets); ++j) { + struct snd_kcontrol_new *kcontrol; + kcontrol = &sun4i_codec_widgets[j]; + if (strcmp(kcontrol->name, new_kcontrol->name) == 0) + memcpy(kcontrol, new_kcontrol, sizeof(*kcontrol)); + } + } + } + scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); if (!scodec) return -ENOMEM;

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[alsa-devel] HDMI Repoll after pm?
by David Henningsson 10 Dec '15

10 Dec '15

Hi Takashi, A recent change in sound.git seems to have introduced some repoll after pm. Is this an intentional change? I don't think it's wrong per se (it stops after 6 repolls), but it breaks my hda-emu tester when "list" does not return the expected input. To reproduce, try running: ./hda-emu codecs/canonical/92hd91bxx-dell-inspiron-5423-ccert-201208-11456 -i 1 > pm .... lots of data .... ** TEST_PM DONE ** HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > list 1: HDMI/DP Jack:0 (1) 2: IEC958 Playback Con Mask:0 (1) 3: IEC958 Playback Pro Mask:0 (1) 4: IEC958 Playback Default:0 (1) 5: IEC958 Playback Switch:0 (1) 6: ELD:0 (1) 7: Playback Channel Map:0 (1) HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > ...notice how the last two lines are somewhat unexpected output for a "list" command, but I can make my tool more robust in case you confirm this is an expected change. // David

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